Re: [asterisk-users] help with DAHDI hangup on calling out.
Jerry Geis wrote: I installed DAHDI (2.1.0.3) on a machine with asterisk 1.4.22 and libpri 1.4.7 and I am getting the error: -- Requested transfer capability: 0x00 - SPEECH -- Called 23/317506 -- Channel 0/23, span 1 got hangup, cause 99 -- Hungup 'DAHDI/23-1' Hangup Cause 99 is: = An Information Element or Parameter Does Not Exist or is Not Implemented. Indicates that the equipment sending this cause has received a message which includes information element(s)/parameter(s) not recognized because the information element identifier(s)/parameter name(s) are not defined or are defined but not implemented by the equipment sending the cause. == I suspect your carrier has not configured your service to permit dialing out. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1-rc4: extension i not working??
sean darcy wrote:
I've have a simple caller id lookup on incoming:
[teliax-in]
..
exten =s,n,GoSub(set-callerid-name,0${CALLERID(num)},1)
[set-callerid-name]
exten = 0,1,NoOp( no CALLERID num set)
exten = 02135590993,1,Set(CALLERID(name)=Matthew )
...
exten = _0!,n,NoOp(CALLERID: ${CALLERID(name)})
exten = _0!,n,Return()
exten = i,1,Return() ; somebody else
Now if there's a callerid that's listed, it all works OK. If there's no
callerid, that works. But if there's an unknown callerid, I'd expect
that to go to the invalid extension - i - and Return(). But look
what happens:
Extension i does not work that way. i is for invalid selections
when using an IVR (well really when using the WaitExten application or
when autofallthru is no or when using Background)
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Re: [asterisk-users] Asterisk quits responding
Chances are you have a DNS problem. Asterisk is trying to look up a hostname and it is not getting a response. Todd Reese wrote: New update: After the console being frozen for about 15 minutes, it just responded to the last reload comand ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk quits responding
Asterisk still does things like use DNS SRV records unless turned off, it also tries to do a DNS lookup for all IP addresses on the system. Update your hosts file and see if that helps. Todd Reese wrote: DNS server rebooted. Everything is back online now. What threw me was that one of the sip provider registries is by ip address and all the internal sip and ata's are by ip. That should have bypassed the DNS. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS text messaging capabilities
Elliot Murdock wrote: Hello! What kind of sms text messaging capabilities does Asterisk have? Asterisk has the ability to use land lines to send SMS messages to a remote device that supports landline SMS. In Europe and much of the rest of the world SMS carriers provide a public PSTN gateway into the mobile phone networks. SMS carriers in the USA do not do this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cut Through DTMF caller ID on SIP phone
Sriram wrote: Setup : Asterisk 1.6 on Fedora Core 9 with TE410P.. 1. I;ve noticed that whenever during background(menu-filename) method - i try to press any key for selection like 1 for some prompt, 2 for another prompt etc...Asterisk takes a while before it takes me to the respective option..Is that normal behaviour ? by the time the caller waits to listen to the appropriate prompt on selecting 1 - he thinks nothing is happening for 2-3 seconds .. fyi, I used to use Trixbox prev. and didnt find any such problem ... This typically happens when you have overlapping extensions. i.e. a menu option 1 and extensions starting with 1. Don't forget to look in include'd contexts, and don't forget wildcards. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem configuring Asterisk as client for ekiga.net -- NAT problem
obitori junk wrote: I am experiencing a 606 not Acceptable error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. In my experience Not Acceptable errors happen because the two endpoints cannot agree on a codec. Try allowing all the codecs in your softphone and in Asterisk sip.conf [general] do a disallow=all and an allow=ulaw.I suggest you do this in [general] when testing because it can sometimes be hard to make sure that a peer/friend/user entry is actually matching the incoming call. Once you get it working you can refine it. You could be having a NAT issue too, but I don't think it is related to your 606 error. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Increase DTMF Tone Duration
Andres wrote: We are running 1.4.22 and have been experiencing problems with certain IVRs and DTMF Tone duration. We would like to be able to increase DTMF Tone duration by 50 to 100ms over what the user is pressing on his phone. We have a PRI test circuit and an analyer in between to measure tone duration. We have tried setting chan_dahdi.conf parameter 'toneduration', but that does not do anything that we can measure. We have also tried setting channel.c parameter #define AST_DEFAULT_EMULATE_DTMF_DURATION 200 to several different values but none seem to alter DTMF duration at all. Does anybody have a clue where we can hardcode DTMF duration for tones going out of a DAHDI Channel to the PSTN? So far we have to tell customer to 'press the buttons' for a little bit longer and it will work with the IVRs in question, but many complain saying that they don't have to do that with their regular landline or cellphone. I suspect you have a DTMF mode mismatch. If Asterisk is expecting RFC2833 or INFO DTMF and the phones are sending inband DTMF then Asterisk won't detect it and won't regenerate the DTMF (and so toneduration would have no effect). In Asterisk 1.4 and later there are some DTMF debug options, as well as SIP and RTP debug options. You should start out by making sure that Asterisk is detecting the tones as DTMF and not simply passing the raw audio thru. I don't know what specific DTMF and RTP debug commands are in 1.4+ (you should be able to look them up in the CLI), as my customers have chosen to skip 1.4 and go directly to 1.6 once they have become comfortable with it. Good luck with this. DTMF issues can be hell to diagnose and fix sometimes. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
Barry L. Kline wrote: Bill Andersen wrote: In the order in which people normally read text they don't repeat the entire conversation from the beginning each time a question is asked either... Bottom posting is just as bad! ./bill Not when you take the time to properly trim your reply it's not. BK To me top posting is like people talking about SIP Trunks. There is no such thing as a SIP Trunk. There are SIP connections, peers, friends, etc. The term is simply a marketing buzzword to make people that don't know much about VoIP feel all warm and fuzzy about a product. You're not going to be able to make people stop top posting and I'm not going to be able to make people stop using wrong or misleading terms like SIP Trunk. If you try all you are going to do is piss people off and stress yourself out. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dedicated Fax Line
In the USA (maybe other T-1 countries) you can have a channelized T-1. Each channel is assigned signaling just like an analog line, FXS, FXO, EM, etc. I have worked with a carrier in the past that could put FXO channels on a T-1 along with a PRI channels on the same T-1. Andrew Thomas wrote: I can only assume it's a T1 thing - as E1's tend not to have that facility. Oh well, you live and learn :) -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Tim Nelson -- Sent: 16 December 2008 15:08 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] Dedicated Fax Line -- -- I've worked with many providers who are able to do this. In fact, -- we're using such a setup on our office PRI. I'm not sure how they're -- achieving this on their end however... -- -- Tim Nelson -- Systems/Network Support -- Rockbochs Inc. -- (218)727-4332 x105 -- -- - Andrew Thomas a...@datavox.co.uk wrote: -- -- Since when can you segment PRI channels off at the telco end? I -- know -- you could do with DASS - but I'm not aware you can do it with PRI. -- -- -- Andrew Thomas -- Technical Services Manager -- DataVox Ltd -- Saddleworth Business Centre -- Huddersfield Road -- Delph, Oldham -- OL3 5DF -- -- ___ -- -- Bandwidth and Colocation Provided by http://www.api-digital.com -- -- -- asterisk-users mailing list -- To UNSUBSCRIBE or update options visit: -- http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow up on parking
Terry Wilson wrote: On Dec 15, 2008, at 7:05 AM, Mike wrote: Just so I'm clear: there is no way to do what I want short of playing with the underlying code, correct? Yes. I'm working on an issue right now related to parking and noticed that Asterisk completely lies with the verbose statement saying that it will time back out to an extension. There is an if/else that checks a string that will always be set and therefore will never hit the else...which is where the code is that would time back out to an extension as opposed to trying to magically find the original caller and call the channel back. It is fairly complex code in there, so it may take a bit to fix...but I thought I'd let your know that I am working on it, anyway. I saw this in 1.2 as well. I don't know about 1.4, since my customers never used 1.4. Since all parked calls were supposed to be sent to the operator, it was not an issue for my customers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow up on parking
[park-dial] ; app_park adds a priority 1 for us, but due to Asterisk oddities, we still need this Noop exten = _.,1,Noop exten = _.,n,Goto(corporate,3500,1) exten = h,1,Noop Mike wrote: That would help me, but I can't even do that (send all parked calls to anybody) because of the dynamic park-dial context. Regards, Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric ManxPower Wieling Sent: Monday, December 15, 2008 16:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Follow up on parking Terry Wilson wrote: On Dec 15, 2008, at 7:05 AM, Mike wrote: Just so I'm clear: there is no way to do what I want short of playing with the underlying code, correct? Yes. I'm working on an issue right now related to parking and noticed that Asterisk completely lies with the verbose statement saying that it will time back out to an extension. There is an if/else that checks a string that will always be set and therefore will never hit the else...which is where the code is that would time back out to an extension as opposed to trying to magically find the original caller and call the channel back. It is fairly complex code in there, so it may take a bit to fix...but I thought I'd let your know that I am working on it, anyway. I saw this in 1.2 as well. I don't know about 1.4, since my customers never used 1.4. Since all parked calls were supposed to be sent to the operator, it was not an issue for my customers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ring back tone
No, not on FXO ports. On FXO ports Asterisk considers the call answered as soon as dialing is finished. Asterisk has no way to detect when the far end answers when using FXO ports. michel freiha wrote: I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way to play the asterisk ring back tone all the time? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ring back tone
Philipp Kempgen wrote: michel freiha schrieb: I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way to play the asterisk ring back tone all the time? Dial(,,r) ? Much like violence and herding of llamas, the r option to Dial (and the Ringing app) almost never solve the problem they are intended to solve and frequently cause more, usually unforeseen, problems. Just say No! to r. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call to mobiles and it is turn off
Remove the r option to Dial. Bruno Castelo Branco wrote: Hi all When I call to any mobile and the device is power off the asterisk keep ringing and I not able to hear the tradicional message saying this mobile is power off. When I call from a normal analogic line I got the message. Somebody have some suggestion to enable asterisk to identify turn off devices and pass the message to peer? otherwise when somebody call to some mobile always think is ringing and not power off. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parked Extension Variable
Use the docs, Luke.
dev-1*CLI core show application parkandannounce
dev-1*CLI
-= Info about application 'ParkAndAnnounce' =-
[Synopsis]
Park and Announce
[Description]
ParkAndAnnounce(announce:template,timeout,dial[,return_context]):
Park a call into the parkinglot and announce the call to another channel.
announce template: Colon-separated list of files to announce. The word
PARKED
will be replaced by a say_digits of the extension in
which
the call is parked.
timeout: Time in seconds before the call returns into the return
context.
dial: The app_dial style resource to call to make the
announcement. Console/dsp calls the console.
return_context:The goto-style label to jump the call back into after
timeout. Default priority+1.
The variable ${PARKEDAT} will contain the parking extension into which the
call was placed. Use with the Local channel to allow the dialplan to make
use of this information.
David Gibbons wrote:
Hello,
When I execute parkandannounce() in the dialplan, is the extension that the
call is parked to stored in a variable? I would like to send it to an AGI
script but can't seem to figure out where the 'announcer' gets its
information.
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Re: [asterisk-users] Asterisk variable for SIP context
Mike wrote:
Sure, that works too, but I needed access to context taken from the sip
entry because I needed to goto(${that_context}), and that context varies
depending on the phone used.
This is when setting __TRANSFER_CONTEXT in my cookie cutter really
big/complexe/database driven extension. I get
__TRANSFER_CONTEX=transfer_context, but once in [transfer_context] I send
the call to the right specific context using a Goto cmd.
Else I would need to replicate that big extension for each of those outgoing
context, which is, from a coding perspective, error prone when modifications
are made later.
Can't you do a Set(ORIG_CONTEXT=${CONTEXT}) or
Set(__ORIG_CONTEXT=${CONTEXT}) (syntax before 1.6, see upgrade.txt for
the 1.6 format) before your Goto.
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Re: [asterisk-users] We think we are cpe but they think they are cpe too
There is a loopback somewhere on the line. Contact your telco and say I see a loopback on the line. Please remove it. Uros Djokic wrote: Hi I have problem with TE121 Digium card. I connected it to modem keymile Music 200 (provided by telco) but I can see 2 red lights on modem (both bellow words rx) and my card is red too. I tried to make experiment and made loopback (pins 1 4 , 2 5) and put it in card and card become green (I hope that is sign card is ok) but on CLI i can see following error message WARNING: We think we are cpe but they think they are cpe too ERROR: got ! frame in state 8 and soon after something like no Dchannel using 16 anyway.Is it normal ? Don't I need to see reconfiguring or reset channel 1 to 30 ? My zaptel.conf and zapata.conf are probably ok because i copied them from another system where everything works fine. I installed asterisk through book on my Ubuntu 8.04 server system.Only thing I didn't install before libpri and zaptel was libtermcap because there is no such package on Ubuntu. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite=yes problem
Reinvites will happen by default. Post your sip.conf [general] and the peers in sip.conf masking only the passwords. Also paste the part of extensions.conf that you use to Dial. BERGANZ François wrote: Now, I have : Client 1 -Asterisk1--Asterisk2 Client 2 I need that sip sign go to Asterisk2 But RTP go to Asterisk1 and no more. Where have I to insert canreinvite ? Thank you -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Eric ManxPower Wieling Envoyé : mercredi 3 décembre 2008 19:25 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] canreinvite=yes problem canreinvite=yes should work as long as 1) there is no NAT involved anywhere in the call path, 2) All legs of the call are using the same codec, 3) you do not have the t/T/w/W (and maybe a few other) options to the Dial line. Remember the only way you can really tell if a reinvite happens is by looking at the RTP audio. The SIP signaling will not and has never had a reinvite feature for signaling. Why did you post the same message at :23, :28, and :35 mins past the hour? If you need immediate support you should contact Digium support and pay for a service contract. BERGANZ François wrote: I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] We think we are cpe but they think they are cpe too
Next time I'll be sure to finish my morning coffee before posting. 8-) Tony Mountifield wrote: In article [EMAIL PROTECTED], Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: There is a loopback somewhere on the line. Contact your telco and say I see a loopback on the line. Please remove it. I don't think this is correct. The OP below said that he put the loopback on himself, as a test. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call parking
Welcome to the world of FreePBX. It would save me quite a bit of time if you could list what ports (port number and signaling) you have on the card and what context you want each port to go into. When I manually merge the two files (after stripping out 37 billion comment lines) I see that channel 1 is defined twice, channel 4 is defined once and and channels 2 and 3 are not defined at all. John covici wrote: Sorry about that -- I think I have things in two places -- they do things in a different order than one would expect. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Packet size limit for HDLC?
ICMP is used to determine maximim packet size. If you or the other end are blocking all ICMP then MTU Path Discovery will not work. It's a classic newbie network admin mistake. Symptoms of this problem would be exactly like you describe. Typically I see this on PPPoE connections. More info: http://www.znep.com/~marcs/mtu/ Roger Schreiter wrote: Hi, I'm using app_pppd with a Digium-PRI-card for PPP connections. I had some strange problems with some IP packets passing and some not, e.g. ftp login went well, but as soon as I tried to up- or download a file, noting was transferred. I finally guessed, it must have to do something with the packet size. Then I started pppd with the parameters mtu 296 and mru 296 as in further times with the analogue modems. Then, everything went fine (for a while). Unfortunately, PPP via ISDN is typically using a MTU and a MRU of 1500, and I found, that some commercial ISDN routers do not allow negotiating MTU and MRU. They insist to use a size of 1500. Since, using CAPI or ISDN4Linux (not via asterisk), pppd is working well with the MTU/MRU value of 1500, I assume, there is some packet size limitation in the asterisk part (including app_pppd). I tried to find any too small buffer or similar, but successless. May I ask you, where do you think, the limitation does come from: - from app_pppd (I don't think so) - from libpri - from chan_dahdi - from the dahdi kernel modules - from the asterisk kernel Any hint is welcome! Regards, Roger. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite=yes problem
canreinvite=yes should work as long as 1) there is no NAT involved anywhere in the call path, 2) All legs of the call are using the same codec, 3) you do not have the t/T/w/W (and maybe a few other) options to the Dial line. Remember the only way you can really tell if a reinvite happens is by looking at the RTP audio. The SIP signaling will not and has never had a reinvite feature for signaling. Why did you post the same message at :23, :28, and :35 mins past the hour? If you need immediate support you should contact Digium support and pay for a service contract. BERGANZ François wrote: I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call parking
John covici wrote: OK, I can park the calls OK, but I don't get the announcement -- I am using freepbx if that makes any difference. If you park a call and do not hear the announcement then you are doing a BLIND transfer, not an ATTENDED transfer. You should be doing attended transfers for parking. If you park a call and hear the announcement and then the hold music then you did not COMPLETE the attended transfer. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call parking
By legacy phone I assume you have an analog card connected to your Asterisk server. I've not used analog phones with Asterisk in many years, but IIRC you need transfer=yes and threewaycalling=yes in zapata.conf/chan_dhadi.conf. You would then do a 2nd flash to complete the transfer. On Polyom phones you do Transfer button/dial number/hear parking slot/Transfer button again/Hang up. John covici wrote: I do the following from the legacy phone: hit theflash and get a dialtone from the call, dial 70, the call is parked, but hangs up from me immediately -- isn't this an attended transfer? on Wednesday 12/03/2008 Eric \ManxPower\ Wieling([EMAIL PROTECTED]) wrote John covici wrote: OK, I can park the calls OK, but I don't get the announcement -- I am using freepbx if that makes any difference. If you park a call and do not hear the announcement then you are doing a BLIND transfer, not an ATTENDED transfer. You should be doing attended transfers for parking. If you park a call and hear the announcement and then the hold music then you did not COMPLETE the attended transfer. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parking calls
It is not a parking solution. Sebastian wrote: Any idea? Please I need advice. Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Sent: lunes, 01 de diciembre de 2008 11:58 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Parking calls Hi, How can I park a call from dialplan and get going?? Example: 1. Answer 2. While follow = false 3. ParkCall 4. Checksomthing à follow = true 5. Endwhile 6. UnParkCall 7. Go on….. The idea is let the call waiting while I do some things on the dialplan, is it possible?? Maybe is not parking the solution?? -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pick up IAX2 calls
You can either add that feature to chan_iax2.c or pay someone to add that feature to chan_iax2.c. Bruno Castelo Branco wrote: Somebody know some work around for it? I still trying to find a solution but nothing seems to work thanks Eric ManxPower Wieling wrote: The problem is that IAX2 does not seem to support call pickup. Bruno Castelo Branco wrote: hi I'm using only IAX extensions and inserted callgroup=1 and callpickup=1 for all IAX extensions in iax.conf. Didn't works for while. thanks Tim Panton wrote: I think it doesn't work across channel types. So it works (if I recall correctly) in IAX or in SIP or in ZAP, but not in mixture. I think that if you have a Dial() that rings several extens, then any of the technologies involved can pickup with *8 So if you have Dial(IAX/fredSIP/billzap/mark) then someone in the same group as fred can pickup with IAX and someone in the same group as bill can pickup with SIP etc. So it's an asterisk thing, not an IAX thing per-se. Tim. P.S. (you could try putting in a dummy 'fred' entry into Dial and iax.conf.) T. On 25 Nov 2008, at 01:09, Bruno Castelo Branco wrote: hi thanks Luis , but doesn't work. For SIP extensions works well *8, but for IAX a tried *8 and ** + iax extension and didn't works Luis Morales wrote: Try with ** + iax extension Regards, Luis Morales On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Somebody knows if pickup call works with IAX2? I enable *8 in features.conf, but doesn't works with IAX2 extensions. Any idea? thanks -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pick up IAX2 calls
The problem is that IAX2 does not seem to support call pickup. Bruno Castelo Branco wrote: hi I'm using only IAX extensions and inserted callgroup=1 and callpickup=1 for all IAX extensions in iax.conf. Didn't works for while. thanks Tim Panton wrote: I think it doesn't work across channel types. So it works (if I recall correctly) in IAX or in SIP or in ZAP, but not in mixture. I think that if you have a Dial() that rings several extens, then any of the technologies involved can pickup with *8 So if you have Dial(IAX/fredSIP/billzap/mark) then someone in the same group as fred can pickup with IAX and someone in the same group as bill can pickup with SIP etc. So it's an asterisk thing, not an IAX thing per-se. Tim. P.S. (you could try putting in a dummy 'fred' entry into Dial and iax.conf.) T. On 25 Nov 2008, at 01:09, Bruno Castelo Branco wrote: hi thanks Luis , but doesn't work. For SIP extensions works well *8, but for IAX a tried *8 and ** + iax extension and didn't works Luis Morales wrote: Try with ** + iax extension Regards, Luis Morales On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Somebody knows if pickup call works with IAX2? I enable *8 in features.conf, but doesn't works with IAX2 extensions. Any idea? thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pick up IAX2 calls
Bruno Castelo Branco wrote: Somebody knows if pickup call works with IAX2? I enable *8 in features.conf, but doesn't works with IAX2 extensions. As I understand it, IAX2 does not support callgroup= and pickupgroup= and *8. This link might be helpful: http://www.freepbx.org/trac/ticket/1568 -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip trunking and call transfer
Maybe because there is no such thing as a SIP trunk, at least in the Asterisk world. Most of us call them peer or friend. The term you are looking for is reinvite. Reinvites allow two devices to send audio directly between the two end points of the call. the SIGNALING stays on the servers, but the audio can be sent directly between the two end points. NAT, transcoding, and the T and t options to dial (as well as other things) will prevent reinvies from happening. nik600 wrote: Maybe my question is not clear or is too stupid? (sorry) Maybe this is already done in SIP trunking? Or (worste case) is impossible to do that? Thanks On Fri, Nov 21, 2008 at 8:53 AM, nik600 [EMAIL PROTECTED] wrote: Hi to all. i-ve got a question: what happen when a call between 2 trunks is transferred to another trunk? For example, suppose that i have 4 trunk A,B,C,D: Caller 1 - Trunk A/B - Caller2 Then Caller 2 transfer to Caller 3 behind Trunk B/C What happend? a) Caller 1 - Trunk A/B - Trunk B/C - Caller3 or b) Caller 1 - Trunk A/C - Caller3 So: is it possible to avoid the scenario a) ? Thanks to all -- /*/ nik600 http://www.kumbe.it -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using MAC or extension number as SIP identifier
Philipp Kempgen wrote: Olivier schrieb: For a long time, I was wondering if I should use MAC address instead of Extension number to identify SIP endpoints (as I'm mostly not using softphones). Before diving into this, I wondered how people using MAC address are using CLI as it seems more natural and simple to type sip show peer 4566 as opposed to sip show peer 00147F784512. Is there something obvious I'm missing (auto-completion ? aliasing ? ...) ? SIP accounts are about users (not extensions or devices). No, SIP accounts are about devices. There is nothing in sip.conf (except maybe the callerid= settings) that tie the information to a specific user. In fact the same user can use multiple devices, all listed in sip.conf. Personally I use the MAC-x wherex=the line appearance number. MAC-a for first line appearance, MAC-b for 2nd, etc. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using MAC or extension number as SIP identifier
Olivier wrote: 2008/11/20 Eric ManxPower Wieling [EMAIL PROTECTED] Personally I use the MAC-x wherex=the line appearance number. MAC-a for first line appearance, MAC-b for 2nd, etc. Is it easy to use (CLI, logs...) ? Would you step back to an extension-based identification scheme ? I would never go back to extension based sip accounts. Some of our users have multiple different extensions per phone and many extensions appear on more than one phone. It is very nice to be able to say to the user read me stuff on the white sticker on the bottom of the phone (which is where Polycom puts the MAC). Then we know EXACTLY which phone the user is talking about, which phone config file, which parts of the dial plan, etc. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 PRI to and from SIP screeching
From IRC: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. Try lowering your rxgain on the PRI. Peter Lindquist wrote: We have just set up trixbox latest with a Rhino r1t1 card, hooked up to a plain E1 PRI line. We call fine SIP to SIP, but as soon as we make a call from SIP to PSTN all sounds become unintelligible screeching or static kind of noise on both ends, when we call PSTN to SIP the PSTN side seemingly OK at least we hear no screeching sound, but the SIP side is a even worse screeching sound. To be certain all is as plain as possible we disabled all echo canceling, and we use only alaw. Any ideas on what this can be and how we start troubleshooting this? -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why Nat=yes Nat=no Option?
Alex Balashov wrote: Steve Totaro wrote: I have done some large installs where people are going to be in the office, sometimes out, work from home, it always changes sorta thing.. I have found that setting all device profiles to Nat=yes Just Works whether they are on the LAN or not and this is even on larger scale systems with hundreds of phones. Is there any reason why this would be frowned upon as a default? Even to the point of, if nat= is not specified, it would default to yes? Is there a performance hit somewhere, or some other downside? If not, I suggest making it the default. The premise of nat=yes is that the domain portion of the Contact URI is overridden with the real, received source IP of the request and that the default expectation of port 5060 (if not specified in the Contact URI) is dropped in favour of the actually received source UDP port. Similarly for SDP (without SIP-aware ALG). I think the reason this would be frowned upon as a default is philosophical in essence; by default, per the RFC, a SIP UAC is expected to behave such and such way, i.e. use the Contact URI that arrives in a REGISTER request and/or INVITE. Overriding that with the received IP:port is a hack around prescribed behaviour, and enabling hacks as default behaviour is generally considered a bad idea. IIRC Uniden phones do not (or did not at one time) work with nat=yes if they were not NAT'd. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get correct dial result for outgoing calls thru ISDN?
Use ${HANGUPCAUSE} which provides a Q.931 hangup cause. chan_sip and
chan_iax (maybe other channels) translate the protocol specific causes
to a Q.931 hangup cause.
I would recommend you check the doc directory in the Asterisk source
code for channelvariables.txt, but for some reason that I can't figure
out most of those docs where translated into TeX format in 1.6. Maybe
the Wiki has a text version.
Daniel - Asterisk wrote:
Hi everyone,
Currectly I'm having some troubles to get correct status of my calls
throug ISDN lines, when outbound calls don't get its destination I
always receive NO ANSWER as ${DIALSTATUS} despite the fact I know the
target number doesn't exists or is busy at that time.
Maybe there is something I must change in my zaptel.conf or zapata.conf,
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Re: [asterisk-users] How to get correct dial result for outgoing calls thru ISDN?
Tilghman Lesher wrote:
On Wednesday 12 November 2008 23:15:01 Eric ManxPower Wieling wrote:
Use ${HANGUPCAUSE} which provides a Q.931 hangup cause. chan_sip and
chan_iax (maybe other channels) translate the protocol specific causes
to a Q.931 hangup cause.
I would recommend you check the doc directory in the Asterisk source
code for channelvariables.txt, but for some reason that I can't figure
out most of those docs where translated into TeX format in 1.6. Maybe
the Wiki has a text version.
If you 'make asterisk.pdf', the LaTeX files format nicely into a singly
contained PDF document.
Don't you need LaTeX installed for that?
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Re: [asterisk-users] Recommend Wireless IP Phone
You can sometimes find the older Cisco Aironet boxes that run at 900Mhz. That frequency is AWESOME in rural areas. Mountains will still block it, but trees and water does not. Drew Gibson wrote: Wilton Helm wrote: Good points. I got an access point instead of a router specifically so I could locate it in the best position. IMO Wi-Fi routers are dumb by definition because where you want a router is probably NOT anywhere close to the best point for the Wi-Fi part. This unit has a particularly sensitive receiver to compliment the higher power. It would have been nice it it had MIMO, too, as that always helps. Repeaters would be a challenge in this case because most of the property is natural wooded (so no power or protection) and I'm trying to cover a road by only own property at one end. naturally wooded does not bode well for WiFi. Trees are much better than walls at absorbing 2.4GHz signals due to their high water content. Mountains block 2.4GHz even better. If the woods are deciduous, it may work well in the winter but fade away come spring. If the road is fairly straight, a directional antenna like a Yagi at one end might give you coverage there. As for the rest of your property, you will have to get an omni antenna up high, say one of your mountains. You may be better off with something that uses lower frequencies. The old analogue cordless phones have much better range than 2.4GHz digital stuff. regards, Drew -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec problems when using G.723
Thomas Winter wrote:
On Sunday 09 November 2008 20:14, Eric ManxPower Wieling wrote:
The best (and maybe only way) is to set your client and your service
provider to only do G.723.
Really, thats not the way it should work.
How I can find out the codec of an incomming call?
Is there any way to use ${SIP_CODEC} to try to change to G.723 and then check
success?
If OK use provider with only allowed G.723 and if not use provider with
allowed alaw and ulaw?
I didn't say that is how it should work. I said that is how it does
work. No, you cannot change the codec of the incoming call in the
dialplan. SIP_CODEC only sets the codec for the outgoing leg of the call.
Remember, Asterisk cannot transcode to/from G.723
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Re: [asterisk-users] Codec problems when using G.723
The best (and maybe only way) is to set your client and your service provider to only do G.723. Thomas Winter wrote: Hi, I have a problem with codecs. I have an provider with allowed codec alaw, ulaw, g.723 I have SIP clients with codec allowed alaw, ulaw, g.723 If a SIP clients wants call through with g.723 Asterisk is using alaw to connect to the provider, so its not working because only passthrough would work. How I can prevent this? best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] twice normal beep before busy tone ??
It sounds like you have analog lines. If that is the case, the silence you experience is Asterisk sending the DTMF down the line. Asterisk collects the DTMF and when you are done dialing it retransmits those digits down the analog line. I think each digit is by default 300ms. If you are dialing 10 digits you have a delay of 3.3 seconds while Asterisk is sending the DTMF. None of the VoIP protocols have his issue. ISDN PRI or BRI also does not have this issue. Stefan Guenther wrote: Matt wrote: -- What this means is that if the call is busy, it will play busy tones, if the call is ringing it will play ringing, congestion, congestion etc. The reason you are hearing silence is that Asterisk doesn't know what the status of the call is before that. The cell phone provider will likely take up to 3 seconds to tell your machine what is happening with the call. If you use the 'r' option then it will play ringing tones even if the phone is busy. well that means, that if I have a bad phone line (meaning poor quality) and I remove the r, I will definitely have silence, when I call cell phone numbers and may have silence on calls to normal phones. If I leave the r in the dial string, this removes the silence and adds the ring tone, with the disadvantage that I will even hear the ring tone on calls to busy numbers. If that is true, the whole problem is related to the quality of the phone line, which prevents asterisk from getting the right status fast enough. Regards, Stefan -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] twice normal beep before busy tone ??
Most IVRs want longer DTMF tone lengths. If you shorten the toneduration= then many IVRs won't work. Wilton Helm wrote: If it is 300 ms, that is way to long. I don't know any CO grade receiver that can't decode in 80 ms and some can do 40. There is also a similar size gap between digits. Is there an option to start dialing as soon as enough digits are collected to guarantee a unique route? That has been the norm in traditional PABXs for 20 or 30 years, and combined with 80 ms duration, it can generally finish by the time the user has entered the last digit. Yes. This feature is called overlap dialing and is only available on PRIs -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question regarding keywords in sip.conf/users.conf
Historically Asterisk's config file parser ignored unknown keywords. This is useful for exactly the things you are trying to do. I hope 1.6 did not remove this feature. Rob Hillis wrote: Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rob Hillis wrote: Unfortunately RealTime isn't going to be an option - it's another level of configuration I want to avoid, but more importantly since I'm planning on being able to run these scripts on an Astlinux install, there won't always be a MySQL database available. If worst comes to worst, and the extra configuration included in sip.conf becomes a problem, I'll move it to another text-based config file - not my preferred option (since I'd like to keep everything close-to-hand) but not a major problem since I'm likely to need a separate config file for global configuration options anyway. You could still store them in sip.conf, just make each line a comment. e.g.: ;[myentry]keyword=value ;[myentry]keyword=value You could then search for ;[myentry] for your keywords and strip them off when you write out the real entries. I did think of that, but the idea of using something that's actually a comment as configuration seems fraught with danger - not to mention it being an awful hack. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk installation
Then you should read the READMEs right now. See the 3 upgrade info files as well as any other READMEs. Christian wrote: Hello, Many thanks for the info. OK, I didn't know that. I just installed it. Usually I read the included read me files and so on but not at this time. But I will be able to use my old zaptel hardware that i used with v1.4? Many thanks, Christian On 2008-10-31 at 22:31 Darrick Hartman wrote: Time for you to discover who's your dahdi... Asterisk 1.6 used dahdi and not zaptel. --Original Message-- From: Christian Sender: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk installation Sent: Oct 31, 2008 5:25 PM Hi all, I've just installed the latest v1.6 release of Asterisk. First, I isntalled libpri. Then i installed zaptel with make config at the end of the isntallation as I usually do. Then I installed Asterisk. However, there is no zapata.conf file in /etc/asterisk. I isntalled the sample configuration files. Any tips? Many thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP # DTMF
core show application dial (this is the official application doc) Pay special attention to the D() option. Rodolfo Alcazar Portillo wrote: Hi. In creating a custom extension, and dialing SIP/222/333#444, seems the party receives only 333 What should I do to send the # symbol? or better, where can I find that syntax? Googled a lot, nothing. Thanks! -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dealing with progress codes
From zapata.conf.sample:
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn't seem to work
; with all telcos.
;
; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
; inband: Signal Busy/Congestion using in-band tones
;
; priindication = outofband
arkda wrote:
Thanks for the reply!
I've played around with R to solve this (probably should have mentioned
that), however I wasn't able to make it work. The message is still played
(this message is from the provider). It will move to the next line in the
dialplan, but as soon as users hear the message they hang up.
Since the progress code comes before actual audio is played to the caller
there has to be a way of catching this and dealing with it in the dialplan,
but nothing I've tried so far works.
On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez [EMAIL PROTECTED] wrote:
Try using a R or r on the Dial command, the R option is better for you in
my opinion.
i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
The R option is going to generate a ring tone when the callee indicates
ringing and is going wait for an Answer. As Progress is just for early
media, you wont get that message.
For more info on the Dial command see:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
On Tue, Oct 28, 2008 at 6:56 PM, arkda [EMAIL PROTECTED] wrote:
Some additional information.
I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an
unusual result:
[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical Response)
This occurs about a second after the user hangs up on the error message
being played from the provider. I have a feeling it's trying to execute the
next step in the dialplan but unable since the caller hung up.
Thoughts, criticism, insults all welcome!
On Tue, Oct 28, 2008 at 12:53 PM, arkda [EMAIL PROTECTED] wrote:
Hi,
I've ran into an issue with a PRI provider in a major metropolitan area
that I haven't needed to deal with before and I was hoping someone might
have some insight on how to handle this within the Asterisk dialplan.
At this location users can't always tell if a number is long distance or
not (there are a lot of area codes and prefixes in the vicinity).
Additionally, users are required by the provider to dial the full 10 digit
number even if a call is local since a local call could be for a few
different area codes and prefixes. The problem is the provider requires a 1
in front of the number for long distance calls, but errors out if the call
has a 1 in front and the call is local.
As a result, users are complaining that they are constantly having to
redial with or without the 1. I've tracked down this behavior when a call
fails:
-- Executing [EMAIL PROTECTED]:1] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:2] GotoIf(SIP/user9-b696fb58,
1?5) in new stack
-- Goto (internal,5551515121,5)
-- Executing [EMAIL PROTECTED]:5] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:6] Answer(SIP/user9-b696fb58,
) in new stack
-- Executing [EMAIL PROTECTED]:7] Set(SIP/user9-b696fb58,
CALLERID(num)=555222) in new stack
-- Executing [EMAIL PROTECTED]:8] Set(SIP/user9-b696fb58,
CALLERID(name)=HiThere) in new stack
-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/user9-b696fb58,
--out the pri--) in new stack
-- Executing [EMAIL PROTECTED]:10] Dial(SIP/user9-b696fb58,
Zap/G2/15551515121) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G2/15551515121
-- Zap/22-1 is proceeding passing it to SIP/user9-b696fb58
-- PROGRESS with cause code 31 received
-- Zap/22-1 is making progress passing it to SIP/user9-b696fb58
-- Hungup 'Zap/22-1'
== Spawn extension (internal, 5551515121, 10) exited non-zero on
'SIP/user9-b696fb58'
The above call was a call that is considered local by the provider. The
caller is then redirected to a message (by the provider) saying 'You do not
need to dial a one or zero...' and the message repeats indefinitely.
I'd like to figure out how to handle this in the dial plan so users do
not even know anything happened. To test to see if I could stop the call
progress and reroute it I've tried this so far:
exten = _NX,1,Set(GROUP(default)=dialpool)
exten = _NX,2,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}19]?5)
exten = _NX,3,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED]
)}18]?BLOCK)
exten = _NX,4,NoOp
exten = _NX,5,Set(GROUP(default)=dialpool)
exten = _NX,6,Answer()
exten = _NX,7,Set(CALLERID(num)=${CLR})
exten = _NX,8,Set(CALLERID(name)=HiThere)
exten
Re: [asterisk-users] adding a second extension
ast_request: No channel type registered for ''SIP'
Notice the extra ' in the message.
That is either an error in the error message or you have a an extra ' in
your Dial line. Something like Dial('SIP/
I'm surprised nobody else noticed this.
Stephen Reese wrote:
On Sun, Oct 19, 2008 at 11:21 PM, Juan Rodríguez [EMAIL PROTECTED] wrote:
Stephen:
Your configuration files looks fine. Try from the CLI issuing originate
SIP/101 extension [EMAIL PROTECTED], having the 101 online, then do that
with
originate SIP/102 extension [EMAIL PROTECTED]. See what happens.
If you got a CVS commit, commit again or try installing a release.
http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz (for
download)
Regards,
Juan
I grabbed the latest tarball and installed it.
The extension rings through to 101 and then when I answer it tries to
ring through to 102 but seems to fail.
ns1*CLI originate SIP/101 extension [EMAIL PROTECTED]
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-08245390,
'SIP/102',20) in new stack
[Oct 19 23:41:40] WARNING[20305]: channel.c:3470 ast_request: No
channel type registered for ''SIP'
[Oct 19 23:41:40] WARNING[20305]: app_dial.c:1450 dial_exec_full:
Unable to create channel of type ''SIP' (cause 66 - Channel not
implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/101-08245390, ) in new
stack
== Spawn extension (default, 102, 2) exited non-zero on 'SIP/101-08245390'
The extension rings through to 102 and when I answer the line it
begins to ring line 101.
ns1*CLI originate SIP/102 extension [EMAIL PROTECTED]
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-08249e28,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/101-08244e88 is ringing
-- SIP/vitel-outbound-0825d1e0 is making progress passing it to
SIP/102-08249e28
-- SIP/vitel-outbound-0825d1e0 is ringing
-- SIP/vitel-outbound-0825d1e0 answered SIP/102-08249e28
-- Packet2Packet bridging SIP/102-08249e28 and SIP/vitel-outbound-0825d1e0
== Spawn extension (default, 101, 1) exited non-zero on 'SIP/102-08249e28'
I'm at a loss. Thanks for your help.
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Re: [asterisk-users] Strip prefix
exten = _+X.,1,Goto(${EXTEN:1},1)
michel freiha wrote:
Dear All,
i have the following context defines in etensions.conf:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
exten = _X.,21,Playback(AR_GetGiveToID)
exten = _X.,22,Wait(2)
exten = _X.,23,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,24,Wait(2)
exten = _X.,25,Playback(/tmp/asterisk-recording)
exten = _X.,26,Wait(2)
exten = _X.,27,Hangup
I just need to remove the '+' sign from the dialed number just in case any
user put the '+' as Internationa prefix...Is that possible?How to do that?
--
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Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs
echo canceling systems. Also see http://www.fnords.org/skillslist.html
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Re: [asterisk-users] srv records not being honoured properly
It should be fairly easy to write an AGI script that does the SRV query, do whatever you want with the response, set a channel variable with the results and use that in your dialplan. Brian J. Murrell wrote: On Fri, 2008-10-17 at 10:32 -0500, Andres wrote: Because Asterisk does not support that. Which is just another way of saying Asterisk is broken then. SRV records have requirements for their correct use. If those requirements are ignored, that is a broken implementation. The only thing that Asterisk does is use the first SRV entry First in terms of what was returned, not sorted by priority and weight, right? but it pays no attention to priorities or weights. It does not care about other SRV entries either. Tsk tsk tsk. This is how things have been as long as I can remember. Wonderful. Nothing like half implementing standards. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] srv records not being honoured properly
Brian J. Murrell wrote: On Fri, 2008-10-17 at 11:18 -0500, Eric ManxPower Wieling wrote: It should be fairly easy to write an AGI script that does the SRV query, do whatever you want with the response, set a channel variable with the results and use that in your dialplan. Maybe. If I were an AGI hacker. But really, should I (and every other Asterisk user) have to? If you fight Asterisk's oddities then you will have a depressing and miserable life. If you embrace Asterisk's oddities then you will have a joyous and enlightened life. 8-) I agree that if Asterisk has SRV support it should work in the way expected. The reason Asterisk's SRV support has not been fixed is because nobody with the skills has thought the issue was important enough to fix. Do you know any programming languages? -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching *, + and # in the dialplan
exten = +13129842314,1,Noop(Happy match!) or exten = _+1NXXNXX,1,Noop(Happier match!) Karl Fife wrote: Steve Murphy [EMAIL PROTECTED] wrote: People have voiced this before; but the cut-down version of RE's that the matching algorithms allow are fairly fast, both in the new and the old pattern matching algorithms. Steve Your explanation is clear and it seems like a good design choice to exclude support for regular expressions, but what seems odd (maybe a bug in fact) is the specific exclusion of characters +, # and *. It sounds like you're saying: exten = [0-9*#+].,... is invalid, therefore not a bug, and that only numeric parameters such as: exten = [0-6].,... would be valid. If this is correct could you please explain the proper way to match any extension beginning with + such as '+13129842314' without also matching: -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about echo cancelation
Olivier wrote: 2008/10/10 Eric ManxPower Wieling [EMAIL PROTECTED] All calls with a 2-wire analog piece have echo. You cannot perceive the echo because it happens so fast on non-VoIP connections. On VoIP calls you have significant extra latency while causes you you to perceive the echo. Do you mean generated locally or generated distantly ? I understand that VoIP extra latency sometrimes renders perceivable what was unperceivable before. What suprises me is to hear that media getways filter one-way only : as 2-wires analog devices produce echo, and every phone has 2-wires analog audio, in every call you've got at least 2 sources of echo : one in each endpoint. Where did you hear that media gateways filter one-way only? Any 2-wire analog leg will be a source of echo. Many, many, many calls do not have a 2-wire leg. Think cell/mobile or endpoints with PRI or T-1. Echo must be removed before the call is converted to VoIP -- in your case the Media Gateway is the device that must remove echo. So, if Alice is hearing its own voice, 1. where does it most probably come from ? 2. where should it be removed ? For both, I would reply : 1. it most probably comes from Bob's phone (as other devices in-between are digital so voice can't leak from there), 2. Alice voice echo should canceled at every location: Bob's PBX, PSTN network (ISDN in the case I had in mind) and Alice's Media gateway If you (Alice) are hearing echo then the echo canceling can be done any time after it leaves Bob's 2-wire circuit but before the audio is converted to VoIP on your end. Telcos echo cancel cell/mobile phone calls (also a high latency path) and long distance calls, but almost never do EC on local calls. This is why you seldom get echo when calling a mobile phone or a long distance number -- you mostly get it on local calls. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1 second delay when connecting calls
Try setting canreinvite=no in each of the device sections on a couple of phones, reload chan_sip.so and see if that fixes things. It has fixed the issue when I've tried it. [EMAIL PROTECTED] wrote: Hello, We are using asterisk 1.6, sangoma pri card, and Cisco 7960 phones. When we make or receive calls there is a delay before voice is heard. Anyone have any ideas on where to start to debug or has anyone seen this before. We have played with settings on pri, asterisk, and phones with no change. Thanks for your help and ideas in advance. Neal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about echo cancelation
Handsets use a 4-wire connection. Handsets with the the volume turned up could cause a form of echo as the microphone picks up the ear piece audio (I call this acoustic echo). Everything I said applies to 2-wire caused echo. Other types of echo is fairly uncommon and cannot be solved by normal echo canceling systems. Most echo canceling systems I've seen (mostly tellabs) only cancel echo in one direction. I suspect all of Digium's EC systems only do echo canceling in one direction as well. Olivier wrote: 2008/10/11 Eric ManxPower Wieling [EMAIL PROTECTED] Olivier wrote: 2008/10/10 Eric ManxPower Wieling [EMAIL PROTECTED] All calls with a 2-wire analog piece have echo. You cannot perceive the echo because it happens so fast on non-VoIP connections. On VoIP calls you have significant extra latency while causes you you to perceive the echo. Do you mean generated locally or generated distantly ? I understand that VoIP extra latency sometrimes renders perceivable what was unperceivable before. What suprises me is to hear that media getways filter one-way only : as 2-wires analog devices produce echo, and every phone has 2-wires analog audio, in every call you've got at least 2 sources of echo : one in each endpoint. Where did you hear that media gateways filter one-way only? From a media gateway vendor (mentioning its own products capabilities). That's the main reason I opened this thread as it surprised me a bit ... Any 2-wire analog leg will be a source of echo. Many, many, many calls do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got event 17 (Polarity Reversal)...
You should not get that message on analog lines in the USA or Canada. I suspect your line has a provisioning issue or is using different signaling than you think it is using. Jim Duda wrote: Can anyone tell me what this message means? Got event 17 (Polarity Reversal)... I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. It appears that I get this Polarity Reversal each time an inbound call hangs up. This results in another ring, but no one is there. It appears as an unknown caller, but I believe its a phantom. Thanks, Jim [Oct 10 12:47:54] NOTICE[6669]: chan_dahdi.c:7379 mwi_thread: Got event 17 (Polarity Reversal)... Passing along to ss_thread -- Starting simple switch on 'DAHDI/4-1' [Oct 10 12:47:55] NOTICE[6671]: chan_dahdi.c:7114 ss_thread: Got event 4 (Alarm)... [Oct 10 12:47:55] NOTICE[6671]: chan_dahdi.c:7114 ss_thread: Got event 17 (Polarity Reversal)... [Oct 10 12:47:56] NOTICE[6671]: chan_dahdi.c:7114 ss_thread: Got event 5 (No more alarm)... -- Executing [EMAIL PROTECTED]:1] Goto(DAHDI/4-1, incoming-dial,s,1) in new stack ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about echo cancelation
All calls with a 2-wire analog piece have echo. You cannot perceive the echo because it happens so fast on non-VoIP connections. On VoIP calls you have significant extra latency while causes you you to perceive the echo. Echo must be removed before the call is converted to VoIP -- in your case the Media Gateway is the device that must remove echo. Olivier wrote: Hi, I'm using the following setup : Alice IPPhone --LAN- Media gateway PSTN --- Phone Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably, when users complains about earing their own voice, that means that distant phone or nearby equipment is leaking : Bob's phone is sending Alice's voice signal back to Alice. So, to properly cancel, I would say Media gateway should substract from incoming signal the signal that left the media gateway few ms before. Discussing here and there, some say that Media Gateway never work this way : it would only filters out locally generated echo. Do you agree with that ? If positive, then what can you do, if Bob's phone generate much echo ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED]
Olivier wrote: I don't have any spare zaptel enabled system I could try this on, but I was not aware of this CHANNEL variable. Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables Maybe, I will add a line in www.voip-info.org http://www.voip-info.org to keep others (me?) from searching again. You should have looked in /path/to/arc/asterisk/doc/channelvariables.txt There's lots of cool information there, and all of it is up to date for your version of Asterisk, unlike voip-info.org. I often wonder why nobody seems to read the docs that are included with Asterisk. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
The most reliable ATA is a channel bank. Vieri wrote: --- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote: Why not swap it all with just IP phone? That's because we have almost 400 analog phones already wired in our building (which is very large). So we need to take advantage of the wiring. Also, if we were to convert to an all-IP phone system (non-ATA), we would need to buy more ethernet switches (currently they're all full) and tunnel cables thtough ceilings and walls. In other words, it would cost a lot more than to simply buy ATAs. What I'm looking for however are STABLE, RELIABLE ATAs... -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED]
You would want three pages, 1.2 docs, 1.4 docs, and 1.6 docs. Mark Hamilton wrote: I don't see why not, Voip-info is very outdated in most respects. Most of it with bad examples, dating to Asterisk 1.x era. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josiah Bryan Sent: September 29, 2008 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED] So, should we (I can do it, if desired) write a script that polls subversion docs directory and imports it into voip-info.org when the the docs are changed? I'd be glad to write and host such a script if the community desires the feature. -josiah SIP wrote: Eric ManxPower Wieling wrote: Olivier wrote: I don't have any spare zaptel enabled system I could try this on, but I was not aware of this CHANNEL variable. Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables Maybe, I will add a line in www.voip-info.org http://www.voip-info.org to keep others (me?) from searching again. You should have looked in /path/to/arc/asterisk/doc/channelvariables.txt There's lots of cool information there, and all of it is up to date for your version of Asterisk, unlike voip-info.org. I often wonder why nobody seems to read the docs that are included with Asterisk. Web and/or context-searchable documentation will ALWAYS win out over a somewhat loose collection of text files. That's basic UI psychology 101. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with pickup extension *8 from features.conf using IAX
I believe chan_iax2 does not support call pickup. Search the archives.
Shazaum wrote:
already tested with an exten?
ex:
exten = _*8.,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _*8.,n,Hangup()
2008/9/27 coco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hello list
I am trying to configure a PBX using Asterisk.
The problem I am havong is the following: I want to use the *8 from
features.conf to pickup any ringing extension from a group, becouse
I want to put the users in call queues and I want anybody from the
company to be able to pick a ringing channel, even if is in a queue.
Whwn using Sip protocol for the users, everithing is going fine, I
can pickup any ringing extension from the group using *8.
But the problem appears when I am using IAX protocol. When issuing
*8 from the IAX phone, asterisk tryes to find the *8 in the dialling
rules returning:
*CLI -- Registered IAX2 '40' (AUTHENTICATED) at 10.0.0.30:4569
http://10.0.0.30:4569
[Sep 27 12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process:
Rejected connect attempt from 10.0.0.30 http://10.0.0.30, request
'[EMAIL PROTECTED]' mailto:[EMAIL PROTECTED] does not exist
This I think is wrong, is something like asterisk cannot read from
features.
With the same setting, when using SIP, i get:
*CLI == Using SIP RTP CoS mark 5
[Sep 27 12:06:23] NOTICE[19802]: chan_sip.c:17092
handle_request_invite: Nothing to pick up for
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
and it works ok.
I am wondering if any had this problem before and if you can help me
figure it out(how to make it work--or if is a bug), or find a
sollution using the app pickup.
I tryed using asterisk 1.4.13, asterisk 1.4.21.2 http://1.4.21.2,
asterisk 1.6-rc6 and always the same problem ocurs.
Regards,
Cosmin
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Re: [asterisk-users] extension definition
This is done in sip.conf, iax.conf, etc, not in extensions.conf. By the time a call gets to extensions.conf it must already be authenticated. Assume the username is robertdobbs and the ip is 209.17.71.61 In sip.conf you would have something like this: [robertdobbs] deny=0.0.0.0/0 permit=209.17.71.61 rest of the options here michel freiha wrote: Hi all, I need please the exact extension definition under extensions.conf that accepts any call coming from an appropriate username and Ip address...This mean that the authentication should be done on username and IP address Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium training course
Tilghman Lesher wrote: On Wednesday 17 September 2008 21:42:20 Steve Totaro wrote: On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL PROTECTED] wrote: Anybody knows how to get a Coupon Code for the discount on the Asterisk training classes??? I am interested on taking that upcoming Asterisk Advance course, and 3K is kinda steep and considering I am still a college student paying this training out of my pocket, every bit helps. Sorry to thread jack. For that matter, I think old timers like myself should automatically get a dCAP. Six or seven years of Asterisk extensive experience should grandfather the dCAP and maybe even the training. I am sure I have a few tricks up my sleeve that the instructors don't know. If memory serves me correctly, there was talk about this very issue when the training and dCAP track came out. I will google it later. Nobody, including Mark Spencer and myself, have gotten a free pass on the dCAP. That said, I think you may be able to take the test for cheap. I don't know the exact price, but in any bootcamp, they allow people to come down on the last day and take only the test, as I did, if they have the space and resources (specifically, for the practical portion). I also used to think the old timers should get grandfathered dCAP cert. Then I looked at some of the stuff the dCAP certification tests for and realized that almost nobody out there that learned Asterisk from the docs and use it in a real world install would be able to pass the dCAP. As Tilghman said you can take JUST the dCAP test for a reasonable fee without having to take the classes. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle
Where did you hear this? Shaun Wingrin wrote: I have heard it said that, Asterisk falls over at 100 simultaneous calls. Is this true? -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SCCP port numbers used for audio stram?
SCCP (aka Skinny), H323, MGCP, and SIP all use the RTP protocol for audio. For all signalling protocols (except maybe H323) use rtp.conf for the RTP ports. OCG Technical Support wrote: I have a 7921 wireless phone working with Asterisk, and I want to tighten the wide open port range of my IPTABLES now. I tried allowing only SCCP port (2000) in/out and found that my audio was gone. A quick look at my iptables message shows source port 15886 and dest port 25968 used: FORWARD - Drop: IN=eth1 OUT=eth2 SRC=172.31.253.4 DST=172.31.254.102 LEN=200 TOS=0x18 PREC=0xA0 TTL=63 ID=0 DF PROTO=UDP SPT=15886 DPT=25968 LEN=180 Can anyone tell my 1. which port range I have to open for the audio stream? 2. Is there a way to force SCCP and the phone to use a different port range for audio? Thanks MD ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX over T1 Question
On a PRI calls come in on ANY B-channel. Therefore you cannot just disable EC on the Fax channels, because there are no dedicated channels for fax. On a Channelized T-1 you can dedicate channels for fax or any other thing. You can't do that on PRI. Steve Totaro wrote: I don't know what Eric is talking about. My advice applies to T1 PRI as does the rest as far as I care to read. Thanks, Steve Totaro On Mon, Sep 8, 2008 at 10:56 AM, Amaru Netapshaak [EMAIL PROTECTED] wrote: Eric, So what is practical for a PRI then? Thank you!!! ++Amaru --- On Fri, 9/5/08, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: From: Eric ManxPower Wieling [EMAIL PROTECTED] Subject: Re: [asterisk-users] FAX over T1 Question To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, September 5, 2008, 10:04 PM The thing is, you are doing FAX over PRI, not FAX over T-1 (which to me implies Channelized T-1). Seems like most people gave you advice that might apply to a Channelized T-1, but would not apply or be practical for a PRI. Amaru Netapshaak wrote: Bob, I should have added that I have disabled hardware EC on the T1 ports. Here is a sample of my zapata.conf -- channels 1-23 are my incoming PRI. This PRI handles both Voice AND FAX calls. Having the hardware EC disabled makes for poor voice communications, and im looking for a way to enable/disable EC per the call type. I understand that echoncancelwhenbridged and Zapata should be telling my A104d to enable/disable HWEC automatically. Channel 25 is the first FXS port on my Rhino CB. It has a FAX directly attached to it. [channels] language= en switchtype = national signalling = pri_cpe pridialplan = national prilocaldialplan= national faxdetect = both echotraining= no echocancel = 256 echocancelwhenbridged = no relaxdtmf = yes overlapdial = no usecallingpres = yes amaflags= default context = default group = 1 channel = 1-23 signalling = fxo_ls faxdetect = both echotraining= no echocancel = no echocancelwhenbridged = no relaxdtmf = yes context = default callerid= FAX 111 channel = 25 Thanks for your assistance everyone! --- On Fri, 9/5/08, Bob Pierce [EMAIL PROTECTED] wrote: From: Bob Pierce [EMAIL PROTECTED] Subject: Re: [asterisk-users] FAX over T1 Question To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, September 5, 2008, 4:43 PM On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote: I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am running Asterisk 1.4.21.2 I think you're mostly right on this setup, but I wonder if your A104d is doing some hardware echo cancellation on these calls. If I'm not mistaken, that can mess up fax machine communications. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
Re: [asterisk-users] SIP Extension Config Issue
You're joking, right? exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr) exten = _1xx,n,Voicemail([EMAIL PROTECTED]) Use whatever voice mailbox and voicemail context you want. Joseph L. Casale wrote: I have a setup with a SIP DID inbound, and several SIP phones inside. Obviously if the SIP phones are off/unplugged/otherwise not available, incoming calls ring busy. My extensions.conf looks like this for inbound calls: exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr) So what could I do to send the call to voicemail if none of the extensions are online? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX over T1 Question
If I am not mistaken every single echo canceler out there will disable itself if it detects a fax tone. Echo Cancelers do not screw up faxes, people screw up faxes. 8-) Bob Pierce wrote: On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote: I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am running Asterisk 1.4.21.2 I think you're mostly right on this setup, but I wonder if your A104d is doing some hardware echo cancellation on these calls. If I'm not mistaken, that can mess up fax machine communications. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX over T1 Question
The thing is, you are doing FAX over PRI, not FAX over T-1 (which to me implies Channelized T-1). Seems like most people gave you advice that might apply to a Channelized T-1, but would not apply or be practical for a PRI. Amaru Netapshaak wrote: Bob, I should have added that I have disabled hardware EC on the T1 ports. Here is a sample of my zapata.conf -- channels 1-23 are my incoming PRI. This PRI handles both Voice AND FAX calls. Having the hardware EC disabled makes for poor voice communications, and im looking for a way to enable/disable EC per the call type. I understand that echoncancelwhenbridged and Zapata should be telling my A104d to enable/disable HWEC automatically. Channel 25 is the first FXS port on my Rhino CB. It has a FAX directly attached to it. [channels] language= en switchtype = national signalling = pri_cpe pridialplan = national prilocaldialplan= national faxdetect = both echotraining= no echocancel = 256 echocancelwhenbridged = no relaxdtmf = yes overlapdial = no usecallingpres = yes amaflags= default context = default group = 1 channel = 1-23 signalling = fxo_ls faxdetect = both echotraining= no echocancel = no echocancelwhenbridged = no relaxdtmf = yes context = default callerid= FAX 111 channel = 25 Thanks for your assistance everyone! --- On Fri, 9/5/08, Bob Pierce [EMAIL PROTECTED] wrote: From: Bob Pierce [EMAIL PROTECTED] Subject: Re: [asterisk-users] FAX over T1 Question To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, September 5, 2008, 4:43 PM On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote: I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am running Asterisk 1.4.21.2 I think you're mostly right on this setup, but I wonder if your A104d is doing some hardware echo cancellation on these calls. If I'm not mistaken, that can mess up fax machine communications. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ringback when the channel is answered
It will do so by default if you have a valid /etc/asterisk/indications.conf (only used for inband tones like after an Answer()) eng. Anatoli Marinov wrote: Hi guys, I am trying to configure an asterisk server for our office. Asterisk 1.4.17 SIP only The problem appears when the call comes from external point to our internal network. So when the server receives the call the channel is answered and the remote user hears prompt which invite him to enter internal private number. After that the server starts to wait the extension. After timeout the server executes Dial application and sends invite to sip client from our internal network. The problem is in this point. I want to play ringback tone to remote user when he waits internal user to pick up his phone but I could not instruct Asterisk to generate fake ringback in rtp stream . -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ringback when the channel is answered
This has nothing to do with the progressinband setting and you should never use the r option. eng. Anatoli Marinov wrote: Is there any special option which I should enable to activate these tones? My progressinband is yes and I cal Dial app with r option it it right? 2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]: It will do so by default if you have a valid /etc/asterisk/indications.conf (only used for inband tones like after an Answer()) eng. Anatoli Marinov wrote: Hi guys, I am trying to configure an asterisk server for our office. Asterisk 1.4.17 SIP only The problem appears when the call comes from external point to our internal network. So when the server receives the call the channel is answered and the remote user hears prompt which invite him to enter internal private number. After that the server starts to wait the extension. After timeout the server executes Dial application and sends invite to sip client from our internal network. The problem is in this point. I want to play ringback tone to remote user when he waits internal user to pick up his phone but I could not instruct Asterisk to generate fake ringback in rtp stream . -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P Card in OFFHOOK state
It would be clearer if it said Hookstate (FXS ports only): Offhook i.e. the state information is not valid for FXO ports. Jay Ray wrote: Any pointers on this one? --- On Tue, 8/26/08, Jay Ray [EMAIL PROTECTED] wrote: From: Jay Ray [EMAIL PROTECTED] Subject: [asterisk-users] X100P Card in OFFHOOK state To: asterisk-users@lists.digium.com Date: Tuesday, August 26, 2008, 12:24 PM After I make a call o n the Zaptel Card X100P FXO moduleit remains offhook state as shown here... Signalling Type: FXS Kewlstart Radio: 0re2uk*CLI Owner: None*CLI Real: Nonek*CLI Callwait: NoneI Threeway: NoneI Confno: -12uk*CLI Propagated Conference: -1 Real in conference: 0 DSP: noore2uk*CLI Relax DTMF: noCLI Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: noLI Pulse phone: noLI Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook -- Sometimes it still takes a new call while in this state and sometimes rejects it... How to correct it such that after I hangup a call it goes back to onhook state... reloading wcfxo module using modprobe clears the issue ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] set callerid with plus sign
+ is not a valid Caller*ID character. Asterisk allows you to use + in Caller*ID, but many carriers will reject the call if you do that. Benny Amorsen wrote: ronald [EMAIL PROTECTED] writes: Is it possible to assign a plus sign on the callerid(num) ? Yes. currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing bs523450017 instead of +6523450017. Which techology? SIP? PRI? POTS? ...? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 5 min limitation on phone calls! how to!
Steve Totaro wrote: On Thu, Aug 21, 2008 at 12:50 PM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote: Hello all! my last month's phone bill sky rocketed after i setup asterisk with softphones all over the house! could someone help me set up a limitation for my wife and kids not to be able to talk for more than 5 min at a time! or like 20 min per week! or whtever limitation i could set for this! any help would trull be appreciated:) You could check out ASTPP or ASTCC and give them their own accounts. The L() option of Dial is used for this sort of thing. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Perl AGI defunct process
Your script is not catching SIGHUP, which is what Asterisk uses to tell the AGI the channel went away. Ruddy Gbaguidi wrote: Hi all. I'm using asterisk 1.4.21.2 and when I run ps -ef |grep defunct, I can see a lot of my perl agi still pending there. The channel has been cleaned up in asterisk. I don't have this kind of problem with python or php. I'm using ubuntu ... Anyone has an idea ? I've tried export LD_ASSUME_KERNEL=2.4.1 but after that I fail to even start asterisk. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?
What would on-board NIC be? Jay R. Ashworth wrote: On Wed, Aug 13, 2008 at 11:54:23PM -0400, Steve Totaro wrote: NIC card is redundant ;-) And you can take that to the ATM machine. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?
But what would you call it? It's not a card, so it can't be a NIC, right? Steve Totaro wrote: Er, it would be one integrated with the MoBo, on the board if you will... Thanks, Steve Totaro On Thu, Aug 14, 2008 at 11:50 AM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: What would on-board NIC be? Jay R. Ashworth wrote: On Wed, Aug 13, 2008 at 11:54:23PM -0400, Steve Totaro wrote: NIC card is redundant ;-) And you can take that to the ATM machine. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?
Steve Totaro wrote: On Wed, Aug 13, 2008 at 7:40 PM, Jonathan Miller [EMAIL PROTECTED] wrote: From what I can determine while troubleshooting a voice-dropping issue, the Asterisk server in my organization has been dropping RTP packets between the asterisk server process and the network interface. [snip] I have ruled the switch out of the problem as it's not seeing the packets on the wire when the issue is occuring. It sounds hardware specific to me. Is this a new install or a new problem? If it is a new problem, then what has changed? Is the NIC in question onboard? What hardware are you using? Brands, MoBo, NIC, etc... If I were you, I would remove or disable the NIC and stick a tried and true old school 3Com NIC in the server and try that. This is my advice as well. Disable the on-board NIC, remove any other NIC card, replace it with a well known good card. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] re-distributing E1
Yes you should be able to do that with an E-1 (it's called DACS). HOWEVER, you can't do DACS on a PRI, as you would need the D-Channel replicated and you can't do that. If you just want Asterisk to provide PRIs to your users, then that's a different story. Hans Witvliet wrote: Before trying something impossible, and making a fool of my self, i rather ask the list... At my work i've got a single E1-test line, and all the project-leaders are constantly fighting over the use of the line. As it is mere the fact that they just need the E1 as a line, but not the amount of traffic, and the fact that they only needs it for several weeks or months, it is not worthwhile ordereing another line (for them) Can i redistribute the traffic of a single E1 transparantly over several other E1's? Was thinking about purchasing one or two quad E1 cards, andmapping incoming calls on cannel 1-5 to the first slave-E1, 6-10 to the second slave-E1 and so on... Just re-mapping B-channels. Most critical part is, that they should not see the difference between the original E1-line and the redistributed one's (besides the fact that their E1 will never be completely be filled) Is their any problem to expect with signaling (fco/fcx), timing and so on... Finally, i presume i need a couple of modem-pairs incase the line-length gets too long, not? hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
The something is generated by Asterisk at the time the call is created. You should never add it, since you don't control that call instance info. In fact, you should almost never care about the call instance string. The -1 means first instance of a call on this channel, a -2 would be seen in you answer a 2nd call for call waiting. Jay R. Ashworth wrote: On Sat, Jul 26, 2008 at 10:48:54PM +0300, Tzafrir Cohen wrote: Except that that is what Asterisk is giving *us*: -- Local/[EMAIL PROTECTED],1 answered Zap/73-1 -- IAX2/VICIast26-19 answered Zap/73-1 -- Zap/11-1 is ringing -- Zap/11-1 answered SIP/101cathy-0824cda0 Trying to dial to Zap/11-1 instead of to Zap/11 is like trying to SIP/101cathy-0824cda0 installed of dialing to SIP/101cathy So, clearly, I'm not smart enough; precisely what are the semantics of the 'Something' in Technology/Channel-Something? Cheers, -- jra -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX to work on two ports: 4569 and 4570
bilal ghayyad wrote: The reason that I need to do this is: I will have two Asterisk PBX's, and I need both of them to use same Internet (so both of them will be behind NAT under same DSL router), in that case, how I will distinguish on the router the calls that need to be send for box A and the calls that need to send for box B? The calls will be handled based on auth info and/or SOURCE port, which your NAT router would handle just fine. It's similar to how two computers on your network can go to the same web site at the same time. With SIP it CAN be a little more complicated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
Jerry Geis wrote: I dont see any errors in the dialplan while loading. I tried to past the whole log but it was rejected. Again 1.2 works, 1.4 works, no on 1.6 I made no changes to the files. I cant even dialplan show default at this time. It looks like you did not read the UPGRADE files for 1.2, 1.4, 1.6 that should have been included in the source code. If you read that you'll realize that dialplan show command was deprecated in 1.4 and be removed in 1.6. Until your read those files you are going to continue to have strange problems. The thing is that they are not strange problems. They are problems you should expect if you don't read the upgrade notes. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
I sit corrected. He should still be reading the upgrade files. Doug Lytle wrote: Eric ManxPower Wieling wrote: It looks like you did not read the UPGRADE files for 1.2, 1.4, 1.6 that should have been included in the source code. If you read that you'll realize that dialplan show command was deprecated in 1.4 and be removed in 1.6. Until your read those files you are going to continue Eric, I think you're mistaken; show dialplan was depreciated, not dialplan show. Doug -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIf Problem
How about:
exten = _9X,n,Goto(not-parked,s,1)
Doug Lytle wrote:
Everybody,
I have a fall though context that, among other things, tests to see if
someone it trying to pick up a non-existent parked call (Defined from 90
to 99). I have the following:
[not-in-service]
exten = _X.,1,Wait(1)
exten = _X.,n,ResetCDR()
; **
; Check to see if the mis-dialed number was a parking
; slot. If so, jump to the not-parked context
; **
exten = _X.,n,GotoIf($[${EXTEN} = 90]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 91]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 92]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 93]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 94]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 95]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 96]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 97]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 98]?not-parked,s,1)
exten = _X.,n,GotoIf($[${EXTEN} = 99]?not-parked,s,1)
I'd like to move it to just one line, such as:
exten = _X.,n,GotoIf($[${EXTEN} = 9?]?not-parked,s,1)
But, I'm not finding a way to do this. Any suggestions?
Doug
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Re: [asterisk-users] delay when rinigng asterisk
Tell your box to not expect Caller*ID information. You set that with usercallerid=no in /etc/asterisk/zapata.conf Since you are using the Asterisk Appliance you would have to contact Digium for support. Sydney Web Hosting wrote: Hi All, I have just setup an asterisk box (AA50) and all is running well. however when I ring the phone number (analog lines) there seems to be a delay. I'm ringing from my mobile phone - It rings 4 times on my mobile before I can hear it in the office. Any ideas on how to shorten this time? Thanks Dave. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wait pickup
chan_iax2 does not support pickup (callpickup=, pickupgroup= and *8). Enrico Pasqualotto wrote: Hi all, One question I have set in the extensions.conf of my asterisk that all incoming call go in the wait application because I need to not connect the caller but remain in the ringing state. After that the call is on the wait exten for a N second I need from other sip phone to pickup this call. There is a way to pickup a call arrived from IAX to an exten wait(999)? I see that the problem is the channel state, my channel in wait is in LINE IS RING but the pickup appl search for channel in REMOTE RINGING. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
Make the card stop sharing it's IRQ with your IDE controller. Try moving the card to another slot. Asterisk has to send an audio packet every 20ms for VoIP calls. I believe Zaptel expects no more than a few ms of latency. If something is causing a delay, like the IDE controller locking interrupts and doing disk activity then you're not going to get your interrupts serviced fast enough and you will have audio issues. Doug Crompton wrote: I am not sure who all see's this list but I do have a few questions that probably only the developers or somone really in the know of Asterisk could answer. - What is the requirement for timing vs. audio playback in Asterisk. Specifically voicemail and IVR's (Not meetme) - Has this requirment changed in newer versions? This obviously is when using Asterisk with no internal cards. I used Asterisk for several years with a P3 Linux system, NO timing, and it worked flawlessly. Now with this new Pentium Dual core system I do not have the perfect audio I experienced with the less powerful system. I fully know there are MANY variable here. It could be a combination of many things, including the OS (Linux Kernel) etc. BUT I offer this input, Music on Hold works fine. This uses mpg123. So why can this palyback fine and the other wav/gsm audio be choppy? I would gladly switch to a newer Asterisk (using 1.2.29) if someone said this was solved in that version. My system can obviously play (mpg123 - background) audio fine. Why then does Asterisk internal audio not also play well? Doug * Doug Crompton * * Richboro, PA 18954* * 215-431-6307 * ** * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change http port on appliance?
You should contact Digium for support for the Asterisk Appliance. It works totally differently from other Digium products. Fidel Garcia wrote: I just found it at : /ramfs/etc/asterisk/http.conf How do I restart the http service without affecting the phone service? Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fidel Garcia Sent: Wednesday, July 02, 2008 4:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to change http port on appliance? I have the AA50 configured with a public address and I would like to change the default http port (80) to something else for security reasons. I cannot find the httpd.conf file anywhere. Help! Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com No virus found in this incoming message. Checked by AVG. Version: 8.0.134 / Virus Database: 270.4.3/1529 - Release Date: 7/1/2008 7:23 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIfTime Function
If any docs were the cause of this (very important) misconception, maybe the docs could be reworded. Do you remember what caused you to think that context was created automatically? broadband Voice wrote: fc7234153*CLI dialplan show open There is no existence of 'open' context I was under the impression that this was part of the Asterisk default libraries. I will create the context then and also add the include files. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
This will happen if the other side is configured the same as the Asterisk side. i.e. PRI CPU mode on both ends or PRI NET mode on both ends. This can also happen if the line is in loopback mode at the far end. Eve-Ellen Cole wrote: The underscore helped, but didn't resolve the real issue. Now I get the following messages: [Jun 19 13:36:15] WARNING[4288] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jun 19 13:36:16] WARNING[4288] chan_zap.c: No D-channels available! Using Primary channel 48 as D-channel anyway! -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding ;password=foo;method=bar to SIP uri
Asterisk allows you to add custom SIP headers. SER is a *very* powerful SIP proxy. I imagine you should be able to make SER translate those headers into the URI as it routes the SIP packet. Tom Browning wrote: To send calls into a custom SER implementation, I need to be able to add some items to the URI that Asterisk will then send as part of the INVITE Asterisk dial SIP/[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] needs to become Asterisk dial SIP/[EMAIL PROTECTED] mailto:[EMAIL PROTECTED];password=foo;method=bar This is not a registration password. It is a passsword associated with the destination xyz at location abc.com http://abc.com Asterisk 1.4.18.1 http://1.4.18.1 seems to glue the data as part of the hostname and fail to lookup abc.com http://abc.com Is there a way to manipulate the URI that will be sent in the INVITE to accomplish this? -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Canadian Whitepage Listing Capability
Joseph L. Casale wrote: So my SIP Provider states they do not offer the service to list my numbers w/ the Whitepages. We phoned the Whitepages and they said we can't do it, the SIP Provider must? Either one/both of them is/are useless or I must switch SIP providers to one that can get this done. Anyone familiar with this fiasco and can help steer me in the right direction? Any suggestions would be greatly appreciated! I am not aware of any ITSPs (Internet phone companies) that provide white pages listings. They could exist, but I doubt it. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Idiot's question
This should do it, but I've not actually tested it. It is based on a
line from my own dialplan.
_X.,n(entrada),Set(CALLERID(num)=${IF($[${LEN(${CALLERID(num)})} =
0]?00:${CALLERID(num):0:11})})
Venefax wrote:
I have two lines in my dialplan that I wish to make it into only one, and I
fail
X.,n(entrada),Set(CALLERID(num)=${CALLERID(num)}00)
X.,n,Set(CALLERID(num)=${CALLERID(num):0:11})
It means: add '00' to the caller id, and then take the first 11
chars from the left. It aims to detect null caller ids and replace them by
zeros. How can I write this expression in just one line?
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T-1, PRI, Frame Relay, Linux, and network design. Based near
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Re: [asterisk-users] Idiot's Question
Oddly core show function SPRINTF works on my 1.6. SPRINTF function
does not seem to be in 1.2 and I don't have any 1.4 systems.
Venefax wrote:
Believe it or not, I cannot find online a single piece of documentation for
the Asterisk function SPRINTF. This example does not work, for it changes
the caller id.
Set(CALLERID(num)=${SPRINTF(%010lld,0${CALLERID(num)})}),
For instance, if the incoming caller id is 17864335989, I get 0684466805 out
of that function, which is not intended one. To be precise, of the caller
has less than 10 chars, I want to complete it with a string of '0's. If the
caller id is nothing, or empty, I want to replace it with 10 zeroes. I guess
I can figure it out if a link to the documentation of SPRINTF is provided.
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Re: [asterisk-users] SIP call, updated with CID as it becomes available
Answer() is seldom the solution. Rob Hillis wrote: Steve Totaro wrote: If you ever have problems with a call dropping after 30 seconds, Answer() is usually the cause. Answer is the /cause/? Or do you mean it's the solution? -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interoffice phone setup
Asterisk gets very upset if it can't lookup the host name associated with every IP on the system, normally it would use DNS to do this, but since your Internet connection was down it could not do that. You should look at /etc/hosts on the Asterisk machine and make sure that each IP address of the system is listed and a name associated with it. You may have the change the order of the items in /etc/nsswitch to make sure file is consulted before dns. Joseph L. Casale wrote: The exact question pose I must leave for others to answer. However, I recently completed a project that overcomes the situation you describe. I installed a cellular gateway giving me a wireless trunk. If I lose IP connectivity I can route calls out through my cell carrier. Works really well. Appreciate the quick response! What I am concerned about is that there are maybe two problems:) Is that behavior at least normal? I don't want to wait until start of business to find out connectivity is up but phones aren't. Just seems odd. Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax on FXS
IP can be run over many things. Internet, Local LAN, Corporate WAN, VPN, etc Each of these things have different characteristics and so I add them to the list. FaxOverVoiceOverIPOverLAN is something that has a good chance of working, as a LAN tends to have little latency and little jitter, where FaxOverVoiceOverIPOverInternet has neither low latency nor low jitter. How would suggest I indicate global internet instead of IP on a local lan? Matt Watson wrote: Ah, you got me there! Could start throwing in a lot of Over's going down that road :) -- Matt http://www.mattgwatson.ca -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, June 10, 2008 4:10 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Fax on FXS On Mon, Jun 09, 2008 at 06:53:34PM -0400, Matt Watson wrote: On June 9, 2008 01:34:31 pm Eric ManxPower Wieling wrote: You should not expect FaxOverVoiceOverIPOverInternet to work well. If you stick to ulaw codec for the entire call, it might work well enough for your use, but it might not. Just as an FYI - you have too many Over's in your description FaxOverVoiceOverIP would make sense, but seeing as how IP is short for Internet Protocol, saying Internet Protocol Over Internet doesn;t make much sense... Unless you use an openvpn / ipsec tunnel :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 open source codec and sample size
The G729 codec is neither open source, nor is it free, and the license/patent does not make an exception for educational use. The Intel LIBRARIES are free for educational/personal use, but the license for that software says that you still need a license from the G729 patent holder before use. I don't understand why people won't pay $10/channel for a fully licensed, legal, and Asterisk supported G729 codec. Manoj_Rajkarnikar wrote: Greetings. I'm new to the asterisk voip world and I'm currently trying out trixbox 2.6.0.7 on a p4 1.8 GHz box. I've downloaded and used the open source g729 codec from site http://asterisk.hosting.lv/ and is working fine. question here is that this codec sends out a packet every 20ms. Though the speech quality is very good, I also like to try out 30ms sampling size to bring down the overhead payload and reduce bandwidth usage. I've searched for it for a couple days with no indication of how to do it. is it possible to change it. do i have to compile my own codec module.. or some patch to asterisk code?? Please suggest. Thanks a lot. Manoj -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interoffice phone setup
The external DNS server would immediately return with a not found
message. Without internet access you'll have to wait for the timeouts, etc.
Joseph L. Casale wrote:
Asterisk gets very upset if it can't lookup the host name associated
with every IP on the system, normally it would use DNS to do this, but
since your Internet connection was down it could not do that.
So to clarify, it not only needs to resolve FQDN's, but do reverse lookups
on ip's as well? I am not sure I noticed this, as the external dns provider
it was using would have no reverse lookup zones for the internal clients?
On an additional note, I have not been able to get onsite yet, but the ISP
repaired
the physical link and the system started working but the inbound sip provider
rang
busy until I ssh'ed in and did a reload from the asterisk console? I thought
the
system would re register any connections define with a register = every
{n} seconds
on its own? Is there something I can do to force what a reload did
automatically so if the link disappears it repairs itself on its own?
Thanks!
jlc
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Re: [asterisk-users] Fax on FXS
You should not expect FaxOverVoiceOverIPOverInternet to work well. If you stick to ulaw codec for the entire call, it might work well enough for your use, but it might not. John Morey wrote: I've been thinking about something around these lines that I'd like feedback on. What I'd like to d,o if it works, is have a fax machine in St. Louis connected up to my asterisk box in Atlanta via Internet/SIP so that anytime the fax machine in St Louis sends a fax it actually goes out through the asterisk box in Atlanta. Something if I understand it correctly like : Fax-SIP(long distance)-Asterisk-FXO-Customer Fax. Would something like this work? -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not picking up incoming calls from TDM400P
Correct. The previous poster was wrong. Drew Gibson wrote: Nope, didn't help. Doesn't the context declaration come *before* the channel assignment in zapata.conf? It's working that way in our main Asterisk server. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. Drew Gibson wrote: Tilghman Lesher wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. Which ports would you expect to be reversed? 5-8 or 9-10? I have similar settings in my fxotune.conf for a TDM2400P and I'm getting complaints of static on the line that I suspect are related to an overtaxed h/w echo canceller My fxotune.conf:- 13=7,255,251,251,2,255,255,1,255 14=9,254,251,255,2,0,1,0,0 15=9,254,251,255,2,0,1,0,0 16=5,0,0,0,0,0,0,0,0 17=9,254,251,255,2,0,1,0,0 18=9,254,251,255,2,0,1,0,0 19=9,254,251,255,2,0,1,0,0 20=9,254,251,255,2,0,1,0,0 21=9,254,251,255,2,0,1,0,0 22=9,254,251,255,2,0,1,0,0 23=9,254,251,255,2,0,1,0,0 24=9,254,251,255,2,0,1,0,0 regards, Drew -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
