Re: [Asterisk-Users] One More IP Phone for interoperability with Asterisk
> How did you work that out? All the IP phones I can see there are video > phones. If a phone ca H.323, SIP, MGCP and Net2Phone all at the same time ... than this is an indication for the PA168 based software. But yes, you're right, I don't know this for sure. BTW: the phone he was mentioning might be not on the web page. Hong Kong stuff is cheap, but I guess that no video conference phone is 75$ :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One More IP Phone for interoperability with Asterisk
> Got to the voip phone product page at > http://www.eezeephone.com/ezp_frm_productdetails.aspx?product_type=101 > and try to match the pictures, descriptions, and prices. Looks like > the matching-game homework my kindergartner brings home. At the end of this website is the AT-323 that I have here for test. I would'nt use it in production, my reasons are written down at http://www.voip-info.org/tiki-index.php?page=Atron In short: it works, but has an even more cheap touch than the Grandstream. Handling (tiny size of buttons, lots of unimportant buttons) is not good. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to start asterisk sourcecode
> I would like to study the asterisk source code(Program). I dont' know > from which file i've to start reading the code. can anyone helpme. Some quick tips: a) don't send HTML e-mail to public mailing lists b) use the mailing list archive of a list before asking questions. The same question you asked was just asked some hours ago --- and answered. c) when you try to understand ANY program, it's helpful to search for the main() function ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] faxing
BTW, compilation of rxfax with latest CVS-2004-07-29 fails. and Makefile.patch (which is on the site) should be modified as well. gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c In file included from app_rxfax.c:14: ../include/asterisk/lock.h: In function `ast_mutex_init': ../include/asterisk/lock.h:300: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) ../include/asterisk/lock.h:300: (Each undeclared identifier is reported only once ../include/asterisk/lock.h:300: for each function it appears in.) make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/src/install/cvs/2004-07-29/asterisk/apps' make: *** [subdirs] Error 1 On Fri, 2004-07-30 at 05:45, Steve Underwood wrote: > Wrong way around. It is passive mode which is giving trouble. I need to > fix the firewalling. Active mode should be OK right now. > > Regards, > Steve -- Best regards Vlad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *
> Please remember that anything that is given here as "usable solution" > should not only be usable but also useful. Whereas Destar is not. It > just does not do anything. It has a totally blank vanilla interface to > it. Please stop submitting plain wrong information to 8000 users or so. I can't see what kind of reputation you can gain from this. No, the web interface is not blank. Yes, DeStar *DOES* anything. It presents a web interface for configurable items (SIP phones, IAX phones, BRI telco lines, Asterisk options) where you can fil in your values. When you press the SAVE, this get's stored. When you select "Write config" on the right side, it will write out a bunch of /etc/asterisk/*.conf files. Yes, it might be that DeStar does not solve your needs. Yes, you did not contact me in any way saying what you were missing. No, I would not use DeStar in a real-life szenario yet, it's still too unfinished. I use it for testing things out. No, it is not yet for end-users. No, It can't configure anything under the sun, e.g. no T1/E1 stuff. It's easy to add this, but no one did so far and I don't have the hardware. Yes, it evolves when I and/or others invest more time into it. Yes, it follows the release-early-and-often scheme. Hey, even Microsoft releases unfinished software :-) At least my web page says that it's still in development, e.g. the comment in parenthesis on http://www.voip-info.org/tiki-index.php?page=Asterisk+GUI or the second sentence on http://www.voip-info.org/tiki-index.php?page=Asterisk+GUI+DeStar. http://www.holgerschurig.de/destar.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to detect the sent status of the Fax
Hi, I start the fax capabilities of Asterisk, but I don't know how to detect that the sent fax status - complete, error, etc. Any ideas? Best Regards, Miroslav Nachev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk+zaphfc: how to answer incoming calls from a BRI (EuroISDN)?
Hi all, finally yesterday I was able to set up my Asterisk machine and now it works greatly. Assuming I use Asterisk also to manage a BRI via a HFC based ISDN card, I can dial outgoing calls via ISDN but I still can't answer incoming calls from that BRI. I read that special extension 's' is for analog only. What do I have to put in extensions.conf to make Asterisk answer incoming calls? I supposed that it should be something like the following: [default] exten => 0991234567,1,Answer exten => 0991234567,2,Goto(100,1) # transfer the call to extension 100 where 099 is my local area code and 1243567 is the phone number associated with the BRI, but it doesn't work. Do I have to configure something else? Thanks, Alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
Walter Klomp <[EMAIL PROTECTED]> wrote: [...] > However, if I dial-in from the SIP phone to my PSTN and then hang up > my PSTN phone, the call does not get disconnected. This is normal and expected behaviour, at least for POTS lines I've used. When you receive a call on a POTS line, you can't clear it by just hanging up. On a POTS line from BT, you can force-clear an inbound call by hitting recall/hookflash then hanging up at the dialtone. The phone will ring for a few moments and then clear the call. -- > IIRC the USA blew up their international telephone exchange very early in > the war. Was that bomb sponsored by AT&T or Cisco? - Mark Clayton and Tim Clark showing cynicism is alive and well in uk.telecom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk+zaphfc: how to answer incoming calls from a BRI (EuroISDN)?
Don't use Answer/Goto, use Dial() instead. exten=0991234567,1,Dial(SIP/abissoli) This assumes that your telco really sends this "0991234567". AFAIK this is different from telco to telco, some omit the zero, some the areacode, some send only the last 4 digits. When you have an extra contest for this, it's easy to catch them all: [in-pstn] exten=0991234567,1,Dial(SIP/abissoli) exten=991234567,1,Dial(SIP/abissoli) exten=1234567,1,Dial(SIP/abissoli) exten=5567,1,Dial(SIP/abissoli) This assumes that you have context=in-pstn in proper section of your zaptel.conf file. (Actually I just changed DeStar in my CVS to generate this config) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Non standard usage of X100P card.
I have two X100P card in my box. I want to connect regular phone (not the phone line!) to one of thse cards. Does anybody think about the same? I don't really want an expensive solution buying additional card with FXS port, I prefer to make something by myself. It'll be great if somebody can point me to technical materials or show electric scheme of such converter. I believe it should be rather simple. I'm ready to cooperate to make such solution. Thanks, Dmitry. _ The new MSN 8: advanced junk mail protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Non standard usage of X100P card.
I have two X100P card in my box. I want to connect regular phone (not the phone line!) to one of thse cards. Does anybody think about the same? I don't really want an expensive solution buying additional card with FXS port, I prefer to make something by myself. It'll be great if somebody can point me to technical materials or show electric scheme of such converter. I believe it should be rather simple. I'm ready to cooperate to make such solution. Thanks, Dmitry. _ The new MSN 8: advanced junk mail protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sound Problem
Hello there, Ich have some interesint problems with SIP & CAPI. Im routing incoming SIP Calls through CAPI to a telephone @ work, but I cant get any sound. If I call from sip client to CAPI direktly I have sound. If I'm recording to a wave file with asterisk there is some sound. My SIP Provider is sipgate Sascha Growe PS.: Sorry for my bad english. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Non standard usage of X100P card.
On Fri, 30 Jul 2004, Dmitry Sergeev wrote: > I have two X100P card in my box. I want to connect regular phone (not the > phone line!) to one of thse cards. Does anybody think about the same? You need something like this: http://www.vikingelectronics.com/products/linesimulator/product_list.html I don't think it will be "rather simple" to make one - it has to make dialtone, ringing etc etc. How about buying an iaxy (s100i) for is it $80? instead? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Non standard usage of X100P card.
Hi Il ven, 2004-07-30 alle 11:18, Dmitry Sergeev ha scritto: > I have two X100P card in my box. I want to connect regular phone (not the > phone line!) to one of thse cards. Does anybody think about the same? no. is completely different. > I don't really want an expensive solution buying additional card with FXS > port, I prefer to make something by myself. It'll be great if somebody can > point me to technical materials or show electric scheme of such converter. I > believe it should be rather simple. the material needed and the time (assuming that's possible) will be much more that the price of a single fxs card... first of all : you'll need to supply power to the line from the card then you must have a ringer to ring the phone (eh, something like 60/70 volts...) then... you must say the card to activate the ringer when needed ... but perhaps this is the simplest step. it's that worthwhile ? -- Matteo Brancaleoni System Administrator [EMAIL PROTECTED] EspiA Srl - e*solution provider Via Pascoli, 37 20129 Milano - Italy SIP:[EMAIL PROTECTED] Tel. +39 0270633354 Fax. +39 0245487890 IAXTEL: 17005662458 http://www.espia.it ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Non standard usage of X100P card.
The hardware is not capable of generating the signals required by the FXS line. Actually a X100P card is a modem which interfaces a telephone line and manipulates the signals on the line with the zaptel driver. It does nothing with the hardware part on it. I think some of the hardware geeks around here give more suggestions on the same. -Kannaiyan - Original Message - From: "Dmitry Sergeev" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 10:18 AM Subject: [Asterisk-Users] Non standard usage of X100P card. I have two X100P card in my box. I want to connect regular phone (not the phone line!) to one of thse cards. Does anybody think about the same? I don't really want an expensive solution buying additional card with FXS port, I prefer to make something by myself. It'll be great if somebody can point me to technical materials or show electric scheme of such converter. I believe it should be rather simple. I'm ready to cooperate to make such solution. Thanks, Dmitry. _ The new MSN 8: advanced junk mail protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] playing a sound during a call
It is certainly possible to play a sound in a channel when the voice stream is through *. You can check the source code of app_dial.c which contain such information. -Kannaiyan - Original Message - From: "shabanip" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 7:48 AM Subject: Re: [Asterisk-Users] playing a sound during a call ofcourse the sound should be heared by two sides of the call? - shabanip Is there any way to play a sound during a call between two endpoints? -shabanip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help for VoIP Gateway with 2 x FXO & 2 x FXS
Hi, I have MicroNet VoIP Gateway SP5014 with 2 x FXO, 2 x FXS & Ethernet ports. One friend say that the Config Menu of this GW is very similar ot WellTech Config Menu. How to start this GW together with Asterisk? Best Regards, Miroslav Nachev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco PRI CLID outbound fails
Hi... we're using asterisk to send calls over SIP to a cisco with two PRIs. There's really two questions here, one about how to get the cisco to properly authenticate over SIP, and the second about callerid. The second is much more critical... 1. in sip.conf I have a section labeled [a.b.c.d] (the cisco's IP address) but asterisk doesn't end up using that for the cisco's communications. Instead it uses the [general] section. I know this because I had to add a usable context to [general] for it to work. How do I make the cisco at a.b.c.d use a particular part of sip.conf instead of general? (or how do I make chan_sip use [a.b.c.d] instead of [general]) 2. No caller ID shows up on remote side of PRI. through Cisco The inbound calls (PSTN->PRI->cisco->sip->asterisk) work fine, and display CLID great. The outbound calls (sip phone->asterisk->sip->cisco->PRI->PSTN) have an ID set in * (I used NoOp to check it, as well as an explicit SetCallerID) and the ID shows on the Cisco console debug messages but NONE shows up on the PSTN side. We've checked the switch type (NI2) with the carrier, and they say their settings are good. We then moved the PRI to the Zaptel PRI card... and sure enough outbound caller ID works, so we know the carrier wasn't lying to us. That tells me it's a cisco configuration problem. I have been unable to find what's wrong with the cisco config. Any help would be appreciated. Ehud [EMAIL PROTECTED] Here's the console output, my comments in parens. Jul 28 14:47:47 MST: %ISDN-6-CONNECT: Interface Serial1/1:0 is now connected to 520XXX unknown Jul 28 14:48:13 MST: %ISDN-6-CONNECT: Interface Serial1/0:0 is now connected to YYY unknown (XXXis the correct callING phone number. YYY is the correct callED phone number. When gets the call there is no callerid for the call.) (and relevant parts of the config) controller T1 1/0 framing esf linecode b8zs cablelength long 0db pri-group timeslots 1-24 ! controller T1 1/1 framing esf linecode b8zs cablelength long 0db pri-group timeslots 1-24 interface Serial1/0:23 no ip address isdn switch-type primary-ni isdn incoming-voice voice isdn bchan-number-order ascending no cdp enable ! interface Serial1/1:23 no ip address isdn switch-type primary-ni isdn incoming-voice voice isdn bchan-number-order ascending no cdp enable voice-port 1/0:23 echo-cancel coverage 32 ! voice-port 1/1:23 echo-cancel coverage 32 ! ! ! dial-peer voice 1 voip application session destination-pattern 8T session protocol sipv2 session target ipv4:ip.address.goes.here session transport udp dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 2 pots incoming called-number 520... direct-inward-dial ! dial-peer voice 3 pots destination-pattern T port 1/0:23 ! dial-peer voice 4 voip application session destination-pattern 520... session protocol sipv2 session target ipv4:ip.address.goes.here dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 6 pots destination-pattern T port 1/0:23 ! smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Running AGI script on answer.
Hi there, Is there an accepted way of running an AGI script on answering of a channel? Is it even possible? I don't need to execute AGI commands, I just need to know a channel has been answered. Thanks, David. -- "One world, one web, one program" -- Microsoft promotional advert. "Ein Volk, ein Reich, ein Fuehrer" -- Adolf Hitler. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] faxing
On Fri, 2004-07-30 at 03:40, Vladyslav wrote: > BTW, compilation of rxfax with latest CVS-2004-07-29 fails. > and Makefile.patch (which is on the site) should be modified as well. > > gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c > In file included from app_rxfax.c:14: > ../include/asterisk/lock.h: In function `ast_mutex_init': > ../include/asterisk/lock.h:300: `PTHREAD_MUTEX_RECURSIVE' undeclared > (first use in this function) > ../include/asterisk/lock.h:300: (Each undeclared identifier is reported > only once > ../include/asterisk/lock.h:300: for each function it appears in.) > make[1]: *** [app_rxfax.o] Error 1 > make[1]: Leaving directory > `/usr/src/install/cvs/2004-07-29/asterisk/apps' > make: *** [subdirs] Error 1 You need to make sure that _GNU_SOURCE is defined or else you will get this error. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Limit // incoming calls to Queue Agents
> -Original Message- > From: Ryan Courtnage [mailto:[EMAIL PROTECTED] > Sent: Thursday, July 29, 2004 5:06 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Limit // incoming calls to Queue Agents > > > Hello, > > Since outgoinglimit is EOL'd, I've implemented > SetGroup/GetGroupCount to > ensure that SIP clients will only have a single call at any > time. Works > perfectly for simple calls using Dial(). > > I'm now struggling to find a way to similarily limit 2nd > calls to SIP clients > that are Agents, who receive their calls from a Queue(). Is > there any way to > accomplish this (without writing a patch)? > > Thanks > Ryan I would also be very interested in this. My Agents also need to make outbound calls, but * tries to route calls to them even when they are on outbound calls. I ended up disabling call waitingfor those agents, and not allowing the calls to go to voicemail. Thanks, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy config samples
Thanks to Florian and Scott, I cvs the latest and used your instructions. It worked perfect.Digium should include these instructions as an alternative in their guide for IAXy. Thanks again, Eugen --- Scott Petersen <[EMAIL PROTECTED]> wrote: > On Thu, Jul 29, 2004 at 10:21:17PM +1200, Eugen > Cristea wrote: > > Asterisk refuses to register IAXy.I am using the > "IAXY > > Configuration Guide" that comes with the IAXy. > > The guide does not say anything about the > [general] > > section in the iax.conf and the handbook has no > IAXy > > example. > > Any hints? > > Thanks > > > > Did you provision the IAXy to give it a server, > username, password etc? There are two ways to do > this, one is using the provisioning tool mentioned > in the guide the other is to use the provisioning > built in to asterisk. > > There is a config file in more recent builds of > asterisk called iaxprov.conf. In there you put a > template entry such as: > > [default] > port=4569 > server=192.168.1.103 > language=en > codec=ulaw > flags=register > tos=lowdelay > > You then add an entry for each IAXy device that you > want to provision and reference this template. Any > entries in the default template will be picked up. > > [backoffice] > user=backoffice > pass=supersecret > template=default > > Once that is set up and you have reloaded the config > files, you can go to the asterisk console and type: > > iax2 provision backoffice > > You will, of course, have to determine what the IP > address of the IAXy is in the first place. This can > be easy or hard depending on if you have access to > the DHCP logs for your network. For more information > on this command you can do: help iax2 provision > > Once you power cycle the IAXy it will attempt to > connect to the asterisk server at 192.168.1.103 and > register with the username and password. This means > that you will have to have an entry in your iax.conf > file that matches. For example: > > [backoffice] > type=friend > host=dynamic > context=default > secret=supersecret > disallow=all > allow=ulaw > > I hope this helps > > Cheers > Scott Petersen > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > Find local movie times and trailers on Yahoo! Movies. http://au.movies.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unauthenticated calls from a specific IP
We put a VWIC and a DSP in a Cisco 1720. The purpose will be for a customer to use a T1 Crossover cable to connect the 1720 into their existing PBX system. It'll be a "Virtual T1 PRI" type of thing. The Cisco 1720 will make the conversion to SIP and send it to our Asterisk server. As far as his PBX is concerned, it's talking to a standard T1 PRI from the local telco or whatever. The issue is Cisco routers don't support SIP registration/authentication. I want this customer to be in his own context in the extensions.conf file. What I was thinking is, if I remove "username" and "secret" from the sip.conf for a standard user entry, but do a "context=whatever and a "host=x.x.x.x" for his specific IP, if an unauthenticated request comes in from that IP it should automatically put him in that context, instead of the default one specific at the top of the file in the [general] section. Also, if I forward several DID's to SIP/customer1 (customer1 being what I put in brackets for this entry, ie [customer1]) it should see the host=x.x.x.x and send it to that IP, regardless of authentication. sip.confExample below: [customer1] context=customer1context type=friend qualify=no host=x.x.x.x canreinvite=no dtmfmode=inband nat=no callerid="Customer 1" <1235551212> accountcode=8785 amaflags=billing insecure=very extensions.conf Example below: [incoming] exten => 1235551212,1,Goto(customer1context,1235551212,1) exten => 1235551213,1,Goto(customer1context,1235551213,1) exten => 1235551214,1,Goto(customer1context,1235551214,1) [customer1context] include => outgoing_local include => outgoing_longdistance include => outgoing_international exten => 1235551212,1,Dial(SIP/customer1,30,r) exten => 1235551213,1,Dial(SIP/customer1,30,r) exten => 1235551214,1,Dial(SIP/customer1,30,r) Maybe I should put a "defaultip=x.x.x.x" in the sip.conf section as well? Will this work? Thanks, Deon P.S - I resubmitted this, with all the activity my previous message may have been overlooked. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0)
Steven, I just would like to apologies for my comment... Steven Critchfield wrote: On Thu, 2004-07-29 at 14:33, Vasyl Rublyov wrote: I really upset from this forum... from last ten posts nothing came back reasonable. None for PRI problem, none for random disconnects I'm sorry free help didn't please you. Find the consultants page and see if you fair better. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoring spool
Hi I have to send 4 outgoing call, all from ZAP1/1 when asterisk start the first call, immediately it start all other calls in the log I see Unable to request channel Zap/1/282 Jul 30 15:05:27 NOTICE[-1242534992]: pbx_spool.c:232 attempt_thread: Call failed to go through, reason 0 What I have to do to make asterisk make calls in sequence ? Thank You Bye Lorenzo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unauthenticated calls from a specific IP
On Fri, 30 Jul 2004 08:56:03 -0400, Deon Rodden <[EMAIL PROTECTED]> wrote: > We put a VWIC and a DSP in a Cisco 1720. The purpose will be for a customer > to use a T1 Crossover cable to connect the 1720 into their existing PBX > system. It'll be a "Virtual T1 PRI" type of thing. The Cisco 1720 will make > the conversion to SIP and send it to our Asterisk server. As far as his PBX > is concerned, it's talking to a standard T1 PRI from the local telco or > whatever. > > The issue is Cisco routers don't support SIP registration/authentication. I > want this customer to be in his own context in the extensions.conf file. Add this line to the cisco section insecure=yes ; To match a peer based by IP address only and not peer and make sure the host=xxx.xxx.xxx.xxx is correct Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Experiance of shipping into the UK
Roy Eddleston wrote: > While someone else is asking, has anybody any experience of > http://www.digitnetworks.com/ > > Are there any other suppliers people would recommend for somebody > importing to the UK? > > Also has anybody in the UK imported any IAXy or OEM X100P - FXO PCI > Cards? I'm interested in knowing what sort of import duty they > attracted, if anybody has the TARIC classification number that would be > useful, I tried to work it out but it's a minefield. We bought a single IAXy for testing from DigitNetworks. They got credit card authorization after about a week and dispatched it after another five or six days. We paid about £30 import duty. Robie. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] D-Link 1120M
Thanks, Turns out to be a DVG-1120M configuration setting. I am using the "LAN" port to connect to my network. It uses the "WAN" address for the RTP setup, setting the NAT address for WAN port to the LAN port resolved the problem. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason A. Kates Sent: Thursday, July 29, 2004 3:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] D-Link 1120M This is my mgcp.conf that works with my D-Link 1120M I hope that it helps. -Jason [general] ;port = 2427 ;bindaddr = 0.0.0.0 [dvg-1120m] host = 10.251.251.253 context = internal-phones canreinvite = no callwaiting=yes cancallforward=yes threewaycalling=yes transfer=yes callgroup=0 pickupgroup=0 callerid="Name 1" mailbox=5000 line => aaln/2 callerid="Name 2" mailbox=4000 line => aaln/1 On Thu, 2004-07-22 at 23:16, Kubat, Philip wrote: > Does anyone have a D-Link 1120M working with asterisk? It can make > and received calls, but there is no audio to the analog phone. There > is audio from the phone. In an ether trace I do not see RTP from > Asterisk to the 1120M, there is RTP from the 1120 to Asterisk. Any > ideas? > > > > Here is my mgcp.conf > > > > [general] > > port = 2427 > > bindaddr = 192.168.1.30 > > > > disallow=all > > allow=ulaw > > > > [dlinkgw] > > host=192.168.1.32 > > context = local > > canreinvite = no > > line => aaln/2 > > line => aaln/1 > > > > -- Jason A. Kates ([EMAIL PROTECTED]) Fax:208-975-1514 IAXtel: 17003723120 FWD:428112 Phone: 212-400-1670 x5000 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing
Hi All.
I connect asterisk and definity by manual at
www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya.
(I just only have E1, not T1 card).
I see, that card work (in definity trunk status, and at asterisk
== D-Channel on span 1 up
-- B-channel 1 successfully restarted on span 1
-- B-channel 2 successfully restarted on span 1
-- B-channel 3 successfully restarted on span 1
-- B-channel 4 successfully restarted on span 1
-- B-channel 5 successfully restarted on span 1
-- B-channel 6 successfully restarted on span 1
-- B-channel 7 successfully restarted on span 1
-- B-channel 8 successfully restarted on span 1
-- B-channel 9 successfully restarted on span 1
-- B-channel 10 successfully restarted on span 1
-- B-channel 11 successfully restarted on span 1
-- B-channel 12 successfully restarted on span 1
-- B-channel 13 successfully restarted on span 1
-- B-channel 14 successfully restarted on span 1
-- B-channel 15 successfully restarted on span 1
(Configured only 15 channels)
Incoming call, from definity is work ok, but when I try outgoing call, I
recive
-- Executing Dial("SIP/1015-870f", "Zap/g1/2073") in new stack
-- Called g1/2073
-- Channel 1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
How fix it?
Configs:
Extensions.conf
exten => 1015,1, Dial(SIP/1015,10,t)
exten => 93261090,1, Dial(Zap/3/93261090)
exten => 2073,1, Dial(Zap/g1/2073)
sip.conf
[1015]
type=friend
secret=phone
host=dynamic
restrictcid=no ; To have the callerid restriced -> sent as ANI
Zapata.conf
[channels]
context=default
switchtype=national
signalling=pri_cpe
group=1
channel => 1-15
/etc/zaptel.conf
defaultzone=us
span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16
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[Asterisk-Users] Softphone - Freeware?!
Hi everybody, What is the most complete Softphone application freeware? X-Lite is very CooL, but the free version don´t support transfers... :( Anyone know, a windows softphone free application that I can use all Askterisk Resources? Congratulations, Jozeph Brasil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Astricon Dev Meeting On Line
> Only the Developer's Meeting will be considered for broadcast > at this time. Why? Seems there are a large number of individuals willing to donate bandwidth and CPU cycles for this. -- Steve Woolley IT Manager ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, Florida 32746 Phone: (407)682-6226 x1110 Fax: (407)682-3455 Cell: (321)229-5311 [EMAIL PROTECTED] www.adstelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Zaptel doesn't see remote hangup ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 - -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello Here is what you can try: Add in zapata.conf: busydetect=yes busycount=6 The maximum it will take for asterisk to see the person hanged-up is after 6 busy dial-tones. Worked for me Jean-Yves On 30/07/2004, at 12:09 PM, Walter Klomp wrote: Thanks Peter, Yes, indeed the problem seems to be exactly what you describe. It's overhere the same. If I dial a mobile number it disconnects immediately when I hangup the mobile. But for analog numbers it takes around 10 seconds or so... Well, at least now I know how to debug pri :-) Walter. - - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 - -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBCkUfXeDVKqIr3GURAoUVAKCJ6NMIMR8MFQTXJVD0vvx7foo5twCeNJBR jZCbw9plXMnAafrgp1Lmqs0= =idgL - -END PGP SIGNATURE- -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBCk/jXeDVKqIr3GURArbMAJ9542GqnGEDcOsYaXKoVd89HmdMPgCfXzUa wmvaNg7sYJxwOzjF3UrbBzk= =pNum -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] permit/deny in sip.conf not working
Hello, I`m trying to make the permit/deny statements work in the sip.conf. They work in denying my phone registration but, still I can just dial a numer and the asterisk will ask the phone for proxy authentocation phone will give the info and * will allow the whole thing to go through. Which kinda defeats the purpose. Is this a bug or am I missing something? The phone is on 192.168.0.0/24 so it should deny it with the config snip I have below. I`m running: CVS-04/22/04-09:13:06 here is a clip from the config: [11] context=local callerid=Anton Yurchenko <11> type=friend username=11 secret=11 mailbox=11 host=dynamic deny=0.0.0.0/0.0.0.0 permit=216.207.245.47/255.255.255.255 Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
We have found X-Lite to be one of the better softphones for Windows. It doesn't support SIP transfers, but it will do the # transfer from within *. Just hit pound during a call and you'll hear Allison say "Transfer", punch in the number you'd like to transfer to and away you go. You do need to have this enabled in the dialplan dial strings to enable transfers. See the Wiki for some help with this. http://www.voip-info.org On Fri, 30 Jul 2004 10:30:12 -0300, Jozeph Brasil <[EMAIL PROTECTED]> wrote: > Hi everybody, > > What is the most complete Softphone application freeware? X-Lite is > very CooL, but the free version don´t support transfers... :( > Anyone know, a windows softphone free application that I can use all > Askterisk Resources? > > Congratulations, > Jozeph Brasil. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco ubr924
Hey list, Does anyone have a current working config example of a cisco ubr924 and * ? I think the 924 only supports MGCP. I want to get VoIP on this device, I was wondering if anyone has already tackled the problem, if not, I'll go in blind :) Thanks Duane Cox
Re: [Asterisk-Users] Softphone - Freeware?!
axra will do. it's an add-on that will give consultative transfer to X-Lite (and others). see below : --- New application for asterisk : axra axra runs separately. developped in C++. it dialogs with asterisk through agi calls and through the manager api. it proccesses phone calls through the dial plan (agi) and concurently through the manager api. axra currently provides consultative transfer for SIP and IAX2 phones. this should easily be extended to any phone technology. hopefully, axra will soon provide 3 way calling. there are two tranfer functions : PreTransfer and CTransfer each should be implemented in the dial plan like : exten => 76,1,AGI(axraagi|PreTransfer) exten => 76,2,Hangup exten => 77,1,AGI(axraagi|CTransfer|auto) exten => 77,2,Hangup you may choose other extensions than 76 & 77. you may omit 'auto' when a call is transfered to PreTransfer (76), the call is parked and waits for a transfer. if the timeout occurs, the call is ringed back. if you call PreTransfer (76) directly, the parked call (if any) is immediatly ringed back. when a call is transfered to CTransfer (77), the call is linked to the pretransfered (parked) call. if no pretransfer exists the call is pretransfered just like 76 was dialed. however, if 'auto' was specified, axra will try to link the call to the oldest live channel attached to transferer's phone. http://www.byortek.com/asterisk/axra-2004-07-29.tgz Please download, read REAME and INSTALL. Any feedback greatly appreciated. - Original Message - From: "Jozeph Brasil" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 3:30 PM Subject: [Asterisk-Users] Softphone - Freeware?! > Hi everybody, > > What is the most complete Softphone application freeware? X-Lite is > very CooL, but the free version don´t support transfers... :( > Anyone know, a windows softphone free application that I can use all > Askterisk Resources? > > Congratulations, > Jozeph Brasil. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
axra will do. it's an add-on that will give consultative transfer to X-Lite (and others). see below : --- New application for asterisk : axra axra runs separately. developped in C++. it dialogs with asterisk through agi calls and through the manager api. it proccesses phone calls through the dial plan (agi) and concurently through the manager api. axra currently provides consultative transfer for SIP and IAX2 phones. this should easily be extended to any phone technology. hopefully, axra will soon provide 3 way calling. there are two tranfer functions : PreTransfer and CTransfer each should be implemented in the dial plan like : exten => 76,1,AGI(axraagi|PreTransfer) exten => 76,2,Hangup exten => 77,1,AGI(axraagi|CTransfer|auto) exten => 77,2,Hangup you may choose other extensions than 76 & 77. you may omit 'auto' when a call is transfered to PreTransfer (76), the call is parked and waits for a transfer. if the timeout occurs, the call is ringed back. if you call PreTransfer (76) directly, the parked call (if any) is immediatly ringed back. when a call is transfered to CTransfer (77), the call is linked to the pretransfered (parked) call. if no pretransfer exists the call is pretransfered just like 76 was dialed. however, if 'auto' was specified, axra will try to link the call to the oldest live channel attached to transferer's phone. http://www.byortek.com/asterisk/axra-2004-07-29.tgz Please download, read REAME and INSTALL. Any feedback greatly appreciated. - Original Message - From: "Jozeph Brasil" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 3:30 PM Subject: [Asterisk-Users] Softphone - Freeware?! > Hi everybody, > > What is the most complete Softphone application freeware? X-Lite is > very CooL, but the free version don´t support transfers... :( > Anyone know, a windows softphone free application that I can use all > Askterisk Resources? > > Congratulations, > Jozeph Brasil. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sccp2 testers needed
Jan: I am willing to test the latest version as I am very interested in some features of my cisco 7960s that are not in the SIP firmware. The only problem is that I am having trouble finding good examples on how to configure the phone / chan_sccp2. I'm used to IAX2 and SIP. I have never touched sccp2/skinny before, but I have been reading everything I can find on it. voip-info wiki and the chan-sccp2 site is sparse in configuration information. If you have some example configs or pointers to more information for a 7960, I would be very greatful if you could pass them along. Thanks, Robert Jan Czmok wrote: Dear Skinny/SCCP lovers :-) I've just completed & uploaded to the cvs the newest version with fixed redial key AND implementation of speed dials. please test extensively and report any bugs. i know that the display is not yet set correctly but the buttons are working as expected. Enjoy testing... --jan (*1) http://chan-sscp.sf.net (*2) yes, bugtracker is down at the moment, will fix this tomorrow morning (in about 8 hours) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Softphone - Freeware?!
Hello, I try to google it without success... :( Sua pesquisa - "enable transfers" site:www.voip-info.org - não encontrou nenhum documento correspondente. Sugestões: - Certifique-se de que todas as palavras estejam escritas corretamente. - Tente palavras-chave diferentes. - Tente palavras-chave mais genéricas. - Tente usar menos palavras-chave. No match! :( -Mensagem original- De: Brian McSpadden [mailto:[EMAIL PROTECTED] Enviada em: sexta-feira, 30 de julho de 2004 10:45 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Softphone - Freeware?! We have found X-Lite to be one of the better softphones for Windows. It doesn't support SIP transfers, but it will do the # transfer from within *. Just hit pound during a call and you'll hear Allison say "Transfer", punch in the number you'd like to transfer to and away you go. You do need to have this enabled in the dialplan dial strings to enable transfers. See the Wiki for some help with this. http://www.voip-info.org On Fri, 30 Jul 2004 10:30:12 -0300, Jozeph Brasil <[EMAIL PROTECTED]> wrote: > Hi everybody, > > What is the most complete Softphone application freeware? X-Lite is > very CooL, but the free version don´t support transfers... :( > Anyone know, a windows softphone free application that I can use all > Askterisk Resources? > > Congratulations, > Jozeph Brasil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
Hi, Have you tried DIAX? It is an IAX2 based soft phone and it is free. Check: http://www.laser.com/dante If you need help, don't hesitate to send me a mail. Best regards, Dan P.S. Use the address from this mail instead of [EMAIL PROTECTED] - Original Message - From: "Jozeph Brasil" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 4:30 PM Subject: [Asterisk-Users] Softphone - Freeware?! Hi everybody, What is the most complete Softphone application freeware? X-Lite is very CooL, but the free version don´t support transfers... :( Anyone know, a windows softphone free application that I can use all Askterisk Resources? Congratulations, Jozeph Brasil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 >You do need to have this enabled in the dialplan dial strings to >enable transfers. u should use something like this: [from-sip] exten => 101,1,Dial(SIP/sip1,20,tTr) from http://www.voip-info.org/wiki-Asterisk+cmd+Dial: The options parameter, which is optional, is a string containging zero or more of the following flags and paramters: t: Allow the called user to transfer the call T: Allow the calling user to transfer the call r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBClf24Q/49nIJTlwRAg/kAJ90/tEQZXEIVe+A1WTM7HDtQGF1dgCeLrCG 01E4lkdvIbjpGrvMoiGu324= =lgdl -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Softphone - Freeware?!
I have one X100P installed with two SIP extensions using X-Lite, I just would like to transfer the call to another SIP extension; Just a "Flash"+"Extension"+"Hangup CALL"... Thanks for all! -Mensagem original- De: Eric Bart [mailto:[EMAIL PROTECTED] Enviada em: sexta-feira, 30 de julho de 2004 10:51 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Softphone - Freeware?! axra will do. it's an add-on that will give consultative transfer to X-Lite (and others). see below : --- New application for asterisk : axra axra runs separately. developped in C++. it dialogs with asterisk through agi calls and through the manager api. it proccesses phone calls through the dial plan (agi) and concurently through the manager api. axra currently provides consultative transfer for SIP and IAX2 phones. this should easily be extended to any phone technology. hopefully, axra will soon provide 3 way calling. there are two tranfer functions : PreTransfer and CTransfer each should be implemented in the dial plan like : exten => 76,1,AGI(axraagi|PreTransfer) exten => 76,2,Hangup exten => 77,1,AGI(axraagi|CTransfer|auto) exten => 77,2,Hangup you may choose other extensions than 76 & 77. you may omit 'auto' when a call is transfered to PreTransfer (76), the call is parked and waits for a transfer. if the timeout occurs, the call is ringed back. if you call PreTransfer (76) directly, the parked call (if any) is immediatly ringed back. when a call is transfered to CTransfer (77), the call is linked to the pretransfered (parked) call. if no pretransfer exists the call is pretransfered just like 76 was dialed. however, if 'auto' was specified, axra will try to link the call to the oldest live channel attached to transferer's phone. http://www.byortek.com/asterisk/axra-2004-07-29.tgz Please download, read REAME and INSTALL. Any feedback greatly appreciated. - Original Message - From: "Jozeph Brasil" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 3:30 PM Subject: [Asterisk-Users] Softphone - Freeware?! > Hi everybody, > > What is the most complete Softphone application freeware? X-Lite is > very CooL, but the free version don´t support transfers... :( > Anyone know, a windows softphone free application that I can use all > Askterisk Resources? > > Congratulations, > Jozeph Brasil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Softphone - Freeware?!
Hi, I can´t connect to your website... :( Is it offline? -Mensagem original- De: Dan [mailto:[EMAIL PROTECTED] Enviada em: sexta-feira, 30 de julho de 2004 11:12 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Softphone - Freeware?! Hi, Have you tried DIAX? It is an IAX2 based soft phone and it is free. Check: http://www.laser.com/dante If you need help, don't hesitate to send me a mail. Best regards, Dan P.S. Use the address from this mail instead of [EMAIL PROTECTED] - Original Message - From: "Jozeph Brasil" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 4:30 PM Subject: [Asterisk-Users] Softphone - Freeware?! Hi everybody, What is the most complete Softphone application freeware? X-Lite is very CooL, but the free version don´t support transfers... :( Anyone know, a windows softphone free application that I can use all Askterisk Resources? Congratulations, Jozeph Brasil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quadbri in NT Mode against PBX.
On Thu, 29 Jul 2004, wendys wrote: > if you connect the Ericcson to the Asterisk you need a crossover-cable. I don't think that's the case when using a QuadBRI card strapped for NT mode. At least we connect an ISDN phone to a QUAD card in NT mode with a straight-through cable and it works. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
Hi, > I can´t connect to your website... :( Is it offline? It works for me. Anyone else with this problem? Try the alternate site at: http://www.geocities.com/tdanro Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] audio delay over time on Zap to SIP
Hello, We have one production server that is identically configured to another production server except for the fact that they each use a different type of digium quad T1 card(one has T400P and one has TE405P). The server with the 405 has been developing delay problems with Sipura SPA-2000 phone adapters after a call has gone on for 15-30 minutes (up to 1 second audio delay). I asked Sipura and they say their adapters have a non-configurable jitter buffer of upto 1 second and that the Sipura is not the cause of the problem. The server with the 400 has identical hardware/network card/Slackware 9.1/Asterisk CVS 2004-07-28/on the same network and has no delay problems with Sipura adapters whatsoever. I have completely wiped the 405 server twice and replaced the network card and the delay always comes back. I am hoping it isn't the 405 card, but that's the only thing that's different from one server to the other. Does anyone have any suggestions? Thanks, MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
Hi! when a call is transfered to PreTransfer (76), the call is parked and waits for a transfer. if the timeout occurs, the call is ringed back. if you call PreTransfer (76) directly, the parked call (if any) is immediatly ringed back. I assume this allows only one PreTransfer event at a time, i.e. things will get messy if we have several users trying to PreTransfer their calls at the same time? Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quadbri in NT Mode against PBX.
This is the difference between the QuadBRI cards and a normal ZapHFC where you use zaphfc drivers. Thomas - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 4:25 PM Subject: Re: [Asterisk-Users] Quadbri in NT Mode against PBX. > > > On Thu, 29 Jul 2004, wendys wrote: > > > if you connect the Ericcson to the Asterisk you need a crossover-cable. > > I don't think that's the case when using a QuadBRI card strapped for NT > mode. > > At least we connect an ISDN phone to a QUAD card in NT mode with a > straight-through cable and it works. > > Steve > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] audio delay over time on Zap to SIP
Matt- Just checking: have you contacted Digium about this problem? I know in the past Mark S referred (on the bug list) to a firmware bug that caused frame sync problems in T1 mode. The problem was with the TE410P, not the TE405P, and was back in Feb, but who knows? I'm not sure where you're located, but I have found that a telephone call to Digium works better than email for things like this. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California & London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Friday, July 30, 2004 7:44 AM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] audio delay over time on Zap to SIP Hello, We have one production server that is identically configured to another production server except for the fact that they each use a different type of digium quad T1 card(one has T400P and one has TE405P). The server with the 405 has been developing delay problems with Sipura SPA-2000 phone adapters after a call has gone on for 15-30 minutes (up to 1 second audio delay). I asked Sipura and they say their adapters have a non-configurable jitter buffer of upto 1 second and that the Sipura is not the cause of the problem. The server with the 400 has identical hardware/network card/Slackware 9.1/Asterisk CVS 2004-07-28/on the same network and has no delay problems with Sipura adapters whatsoever. I have completely wiped the 405 server twice and replaced the network card and the delay always comes back. I am hoping it isn't the 405 card, but that's the only thing that's different from one server to the other. Does anyone have any suggestions? Thanks, MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing *-initiated calls from spool directory not working
I'm running: Asterisk CVS-HEAD-07/06/04-17:49:49 built by [EMAIL PROTECTED] on a i686 running Linux I've tried placing files (both ending in .call and not) in the correct format in /var/spool/asterisk/outgoing I get _nothing_. No log messages, nothing on the console, zip. Permissions seem to be correct on both the files and the directory, as well (* is running as root, for right now during testing). gf-002-pbx-001 root # grep astspooldir /etc/asterisk/asterisk.conf astspooldir => /var/spool/asterisk gf-002-pbx-001 root # grep ^console /etc/asterisk/logger.conf console => notice,warning,error,debug gf-002-pbx-001 root # ls -tla /var/spool/asterisk/outgoing total 12 drwxrwxrwx 2 root root 4096 Jul 30 15:00 . -rw-r--r-- 1 root root 1292 Jul 26 05:08 test.call drwxr-xr-x 5 root root 4096 Jul 25 23:14 .. I've tried mv'ing the files there, tried cp'ing the files there, tried creating them in place... Nothing. I've tried looking around for documentation on this issue, but I just can't find where I'm going wrong. Any suggestions are appreciated. Thanks in advance, -j -- Jeffrey Paul, Senior Network Administrator - [EMAIL PROTECTED] Group Financial LLC / Diamond Financial Products 4000 Town Center/Suite 1000/Office 1013/Southfield/MI/48075-1501 o: 800-476-5882 x244 m: 800-476-5882 x468 f: 800-510-1405 DE2B 3F61 14A1 BD0F B496 DC91 3D97 8C4D 7678 4A42 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] audio delay over time on Zap to SIP
I already have one unanswered ticket open with them right now, guess I'll call them on this one. I had one of the 410's with the sync problem and this is very different from that. One more piece of information to confuse things. I have been in on 5 meetme conferences this morning with a Zap channel a Sipura phone and a Grandstream phone on both servers. And on the 405 server I can actually hear the delay happening to the Sipura through the Grandstream, and it won't happen on the 400 server. And yes the Sipura's have the same firmware(I've tried upgrading that too). And the time that the delay starts is random, some time between 5 minutes and a half hour. It seems to happen faster when the system is under heavy load. I'm going to try swapping motherboards tonight if I can't figure this out. Thanks, MATT--- -Original Message- From: Scott Stingel [mailto:[EMAIL PROTECTED] Sent: Friday, July 30, 2004 10:54 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] audio delay over time on Zap to SIP Matt- Just checking: have you contacted Digium about this problem? I know in the past Mark S referred (on the bug list) to a firmware bug that caused frame sync problems in T1 mode. The problem was with the TE410P, not the TE405P, and was back in Feb, but who knows? I'm not sure where you're located, but I have found that a telephone call to Digium works better than email for things like this. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California & London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Friday, July 30, 2004 7:44 AM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] audio delay over time on Zap to SIP Hello, We have one production server that is identically configured to another production server except for the fact that they each use a different type of digium quad T1 card(one has T400P and one has TE405P). The server with the 405 has been developing delay problems with Sipura SPA-2000 phone adapters after a call has gone on for 15-30 minutes (up to 1 second audio delay). I asked Sipura and they say their adapters have a non-configurable jitter buffer of upto 1 second and that the Sipura is not the cause of the problem. The server with the 400 has identical hardware/network card/Slackware 9.1/Asterisk CVS 2004-07-28/on the same network and has no delay problems with Sipura adapters whatsoever. I have completely wiped the 405 server twice and replaced the network card and the delay always comes back. I am hoping it isn't the 405 card, but that's the only thing that's different from one server to the other. Does anyone have any suggestions? Thanks, MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing
Roman Bessyadovskii wrote:
Hi All.
I connect asterisk and definity by manual at
www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya.
(I just only have E1, not T1 card).
I see, that card work (in definity trunk status, and at asterisk
Incoming call, from definity is work ok, but when I try outgoing call, I
recive
-- Executing Dial("SIP/1015-870f", "Zap/g1/2073") in new stack
-- Called g1/2073
-- Channel 1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
How fix it?
Do you have the "Dial Plan" set up properly
on the Definity side?
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[Asterisk-Users] zaphfc hardware & sound trouble
Hi,
I've been learning asterisk for a couple of weeks now - and it worked for me
as faar as standard configurations where concerned (sip/iax
outbound/isdn4linux & capi with AVM Fritz!, Digium X100P FXO).
Now I recently I'evaluating to use asterisk as a replacemnt for our
companies (15 employees) legacy pbx system and I'm experiencing multiple
problems with the hfc isdn cards:
My evaluation configuration looks like this now:
Software:
# Linux: Mandrake Linux 9.1
Kernel: 2.4.21-0.13mdk
Asterisk: Asterisk CVS-07/25/04-20:08:18 (compiled vom bri-stuff.0.0.2
package von junghanns)
# Hardware:
ISDN: Cologne Chip Designs GmbH ISDN network controller [HFC-P
CI] (rev 2) (Acer PCI Surf) on EUROISDN NTBA in Austria (2 isdn phones, 1
zyxel 2864i isdn modem & 1 Teles ISA ISDN Card on same bus from Telekom
Austria) (mounted in slot 4 (non-shared irq slot)
Mainboard: Vendor: MSI, Product: MS-6712, Version: 1.0 (BIOS Information
Block Vendor: American Megatrends Inc. Version: Version 07.00T
Release: 04/02/01) with AMD Duron 1,2 Ghz
lspci:
00:00.0 Host bridge: VIA Technologies, Inc. VT8366/A/7 [Apollo KT266/A/333]
00:01.0 PCI bridge: VIA Technologies, Inc. VT8366/A/7 [Apollo KT266/A/333
AGP]
00:06.0 Ethernet controller: 3Com Corporation 3c905B 100BaseTX [Cyclone]
(rev 24)
00:08.0 Network controller: Cologne Chip Designs GmbH ISDN network
controller [HFC-PCI] (rev 02)
00:0a.0 VGA compatible controller: Matrox Graphics, Inc. MGA 2064W
[Millennium] (rev 01)
00:0c.0 SCSI storage controller: Adaptec AIC-7861 (rev 01)
00:10.0 USB Controller: VIA Technologies, Inc. USB (rev 80)
00:10.1 USB Controller: VIA Technologies, Inc. USB (rev 80)
00:10.2 USB Controller: VIA Technologies, Inc. USB (rev 80)
00:10.3 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 82)
00:11.0 ISA bridge: VIA Technologies, Inc. VT8235 ISA Bridge
00:11.1 IDE interface: VIA Technologies, Inc. VT82C586/B/686A/B PIPC Bus
Master IDE (rev 06)
00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233 AC97
Audio Controller (rev 50)
00:12.0 Ethernet controller: VIA Technologies, Inc. VT6102 [Rhine-II] (rev
74)
#interrupts:
CPU0
0:5944488IO-APIC-edge timer
1: 1351IO-APIC-edge keyboard
2: 0 XT-PIC cascade
8: 1IO-APIC-edge rtc
9: 0 IO-APIC-level acpi
14:1720179IO-APIC-edge ide0
15: 50IO-APIC-edge ide1
17: 354543 IO-APIC-level eth0
18: 8304 IO-APIC-level aic7xxx
21: 0 IO-APIC-level usb-uhci, usb-uhci, usb-uhci, ehci-hcd
22: 0 IO-APIC-level VIA8233
23:223 IO-APIC-level eth1
NMI: 0
LOC:5944174
ERR: 0
MIS: 0
###zaptel.conf
loadzone = nl
defaultzone = nl
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
###zapata.conf
context=default
switchtype=euroisdn
pridialplan=unknown
overlapdial=yes
signalling=bri_cpe_ptmp
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=yes
context=alex
group=1
channel=2
immediate=yes
echocancel=yes
callerid=asreceived
nationalprefix=0
internationalprefix=00
extension.conf
[alex]
exten => ,1,Dial(SIP/alexander,20,Ttr)
exten => ,2,Voicemail()
exten => ,3,Hangup
[sip]
include => alex
exten => 70,1,Dial(SIP/alexander,30,Ttr)
exten => 70,2,Voicemail,u
exten => 70,3,Hangup
exten => _0.,1,Dial,Zap/g1/${EXTEN:1}
zaphfc modul load syslog messages
Jul 30 14:45:36 faar kernel: Zapata Telephony Interface Registered on major
196
Jul 30 14:45:36 faar kernel: PCI: Enabling device 00:08.0 ( -> 0003)
Jul 30 14:45:36 faar kernel: zaphfc: card configured at mem 0xd1454e00 fifo
0xc2c18000(0x2c18000) IRQ 19 HZ 100
Jul 30 14:45:36 faar kernel: zaphfc: Card 0 configured for TE mode
Jul 30 14:45:36 faar kernel: zaphfc: 1 hfc-pci card(s) in this box.
Jul 30 14:45:36 faar kernel: Registered tone zone 3 (Netherlands)
###/var/log/asterisk/messages
Jul 30 14:48:33 WARNING[16384]: Ignoring port for now
Jul 30 14:48:36 WARNING[163851]: PRI: received TEI check request for TEI =
127
Jul 30 14:48:37 WARNING[163851]: PRI: received TEI check request for TEI =
127
##Problem:
a) when i try to make an inbound call to I get the following message
on the cli prompt
-- Going to extension s|1 because of immediate=yes
-- Extension 's' in context 'alex' from '17109904' does not exist.
Rejecting call on channel 2, span 1
s doesn't exist, that is true, but I would like to receive calls for the msn
38588 an none else, thus the way it worked for capi doesn't seem to work
here
same is true when I dial out - I can't set the outgoing msn, thus my
provider sends out the call as a global call on the bri line
RE: [Asterisk-Users] Outgoing *-initiated calls from spool directory not working
Hi Jeffrey- I use the outgoing call feature all the time and it seems to work ok, but haven't updated CVS in a couple weeks, so maybe there's a new bug. Just for fun, please post the contents of one of your call files. Also, just checking that you find nothing in /var/log/asterisk/messages? Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California & London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeffrey Paul Sent: Friday, July 30, 2004 8:05 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Outgoing *-initiated calls from spool directory not working I'm running: Asterisk CVS-HEAD-07/06/04-17:49:49 built by [EMAIL PROTECTED] on a i686 running Linux I've tried placing files (both ending in .call and not) in the correct format in /var/spool/asterisk/outgoing I get _nothing_. No log messages, nothing on the console, zip. Permissions seem to be correct on both the files and the directory, as well (* is running as root, for right now during testing). gf-002-pbx-001 root # grep astspooldir /etc/asterisk/asterisk.conf astspooldir => /var/spool/asterisk gf-002-pbx-001 root # grep ^console /etc/asterisk/logger.conf console => notice,warning,error,debug gf-002-pbx-001 root # ls -tla /var/spool/asterisk/outgoing total 12 drwxrwxrwx 2 root root 4096 Jul 30 15:00 . -rw-r--r-- 1 root root 1292 Jul 26 05:08 test.call drwxr-xr-x 5 root root 4096 Jul 25 23:14 .. I've tried mv'ing the files there, tried cp'ing the files there, tried creating them in place... Nothing. I've tried looking around for documentation on this issue, but I just can't find where I'm going wrong. Any suggestions are appreciated. Thanks in advance, -j -- Jeffrey Paul, Senior Network Administrator - [EMAIL PROTECTED] Group Financial LLC / Diamond Financial Products 4000 Town Center/Suite 1000/Office 1013/Southfield/MI/48075-1501 o: 800-476-5882 x244 m: 800-476-5882 x468 f: 800-510-1405 DE2B 3F61 14A1 BD0F B496 DC91 3D97 8C4D 7678 4A42 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit // incoming calls to Queue Agents
On July 30, 2004 12:28 pm, Robert Jackson wrote: > > I'm now struggling to find a way to similarily limit 2nd > > calls to SIP clients > > that are Agents, who receive their calls from a Queue(). Is > > there any way to > > accomplish this (without writing a patch)? > > I would also be very interested in this. My Agents also need to make > outbound calls, but * tries to route calls to them even when they are on > outbound calls. I ended up disabling call waitingfor those agents, and > not allowing the calls to go to voicemail. I went ahead and submitted an enhancement request. Please review and add to it if needed: http://bugs.digium.com/bug_view_page.php?bug_id=0002180 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfers- Please Help ASAP
Can someone please help with this. After an outside caller has been parked, they inherit our abilitites to transfer. I have played with all the different combinations of T and t, but nothing seems to work. I found a way to get my Sipura to work with a flash transfer. So right now I am stuck. Is there any way to block callers that have been parked from being able to park us. The only time that this happens is after the call has been parked. Plase help! Im at a loss and people here want answers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing
Yes, I can make a call on that extension from other definity phone, if you
mean it.
-Original Message-
From: Ken Godee [mailto:[EMAIL PROTECTED]
Sent: 30 ÉÀÌÑ 2004 Ç. 19:14
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1.
Incoming work, but not outgoing
Roman Bessyadovskii wrote:
> Hi All.
>
> I connect asterisk and definity by manual at
> www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya.
> (I just only have E1, not T1 card).
>
> I see, that card work (in definity trunk status, and at asterisk
>
> Incoming call, from definity is work ok, but when I try outgoing call, I
> recive
>
> -- Executing Dial("SIP/1015-870f", "Zap/g1/2073") in new stack
> -- Called g1/2073
> -- Channel 1, span 1 got hangup
> -- Hungup 'Zap/1-1'
> == No one is available to answer at this time
>
> How fix it?
Do you have the "Dial Plan" set up properly
on the Definity side?
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[Asterisk-Users] FW: Limit incoming calls to SIP Channels
Hi All, Can someone please tell me how to limit incoming calls to SIP channels using the SetGroup & Checkgroup command. I don't want any call waiting on SIP channels and you are somehow meant to be able to do it with these commands. Many Thanks Daniel Niasoff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0)
On Fri, 2004-07-30 at 08:07, Vasyl Rublyov wrote: > Steven, > > I just would like to apologies for my comment... apology accepted, hopefully for the group. As for your questions. It seems you are experiencing something that there is such a limited pool of people to answer and/or test for. This happens a lot as open source software pushes the boundaries back further. At some point you have to be the leader into an area. So it seems you may have been the leader this time. So like any other time you are a leader, you must push forward and figure out where it leads you. > Steven Critchfield wrote: > > >On Thu, 2004-07-29 at 14:33, Vasyl Rublyov wrote: > > > > > >>I really upset from this forum... from last ten posts nothing came > >>back reasonable. > >>None for PRI problem, none for random disconnects > >> > >> > > > >I'm sorry free help didn't please you. Find the consultants page and see > >if you fair better. > > > > -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc hardware & sound trouble
Alex, I am using zaphfc now. I also use an Epia V 1 mainboard (with Nemiah processor). Try this: 1. asterisk/Makefile MARCH=i586 option Make sure that when you hit "make", actually --march=i586 will be used for an Epia mainboard. On my system it had choosen i686 (with some little bit older CVS head), and this had problems in general with sound. 2. Try running CVS Head + bristuff 0.1.0RC2k (not anything other below i) It fixes problems about audio. -> German: + Rauschen, Knistern in der Leitung! You have this error message: Jul 29 23:46:49 faar kernel: zaphfc: bchan rx fifo not enough bytes to > receive! (z1=3983, z2=3976) It will disappear with bristuff 0.1.0RC2k! 3. How do you use bristuff 0.0.2 but also CVS Head 07/25/04 ? Both don't work together! You have to use CVS Stable if you want to use 0.0.2. 4. Errors a) Try s/MSN instead of MSN in your extensions.conf dial plan. d) see 2. f) That PRI errors seem to be still a major problem with bristuff. I have them all the time, even not the exact error like you. g) link frame errors sound to me not strange, not sure about the exact error Personally I don't see any advantage using a HFC card instead of an AVM with chan_capi. Chan_capi seems to me not under the same development pressure like bristuff and seems already be more stable. It makes no difference using zaphfc or chan_capi if you use it for connecting PSTN (TE). I only use HFC + bristuff for using NT mode, to connect my existing pbx system. ; p2mp NT mode signalling = bri_net_ptmp Chan_capi with AVM card doesn't support that. Greetings Thomas - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 5:15 PM Subject: [Asterisk-Users] zaphfc hardware & sound trouble > Hi, > > I've been learning asterisk for a couple of weeks now - and it worked for me > as faar as standard configurations where concerned (sip/iax > outbound/isdn4linux & capi with AVM Fritz!, Digium X100P FXO). > > Now I recently I'evaluating to use asterisk as a replacemnt for our > companies (15 employees) legacy pbx system and I'm experiencing multiple > problems with the hfc isdn cards: > > My evaluation configuration looks like this now: > > Software: > > # Linux: Mandrake Linux 9.1 > > Kernel: 2.4.21-0.13mdk > > Asterisk: Asterisk CVS-07/25/04-20:08:18 (compiled vom bri-stuff.0.0.2 > package von junghanns) > > # Hardware: > > ISDN: Cologne Chip Designs GmbH ISDN network controller [HFC-P > CI] (rev 2) (Acer PCI Surf) on EUROISDN NTBA in Austria (2 isdn phones, 1 > zyxel 2864i isdn modem & 1 Teles ISA ISDN Card on same bus from Telekom > Austria) (mounted in slot 4 (non-shared irq slot) > > Mainboard: Vendor: MSI, Product: MS-6712, Version: 1.0 (BIOS Information > Block Vendor: American Megatrends Inc. Version: Version 07.00T > Release: 04/02/01) with AMD Duron 1,2 Ghz > > lspci: > > 00:00.0 Host bridge: VIA Technologies, Inc. VT8366/A/7 [Apollo KT266/A/333] > 00:01.0 PCI bridge: VIA Technologies, Inc. VT8366/A/7 [Apollo KT266/A/333 > AGP] > 00:06.0 Ethernet controller: 3Com Corporation 3c905B 100BaseTX [Cyclone] > (rev 24) > 00:08.0 Network controller: Cologne Chip Designs GmbH ISDN network > controller [HFC-PCI] (rev 02) > 00:0a.0 VGA compatible controller: Matrox Graphics, Inc. MGA 2064W > [Millennium] (rev 01) > 00:0c.0 SCSI storage controller: Adaptec AIC-7861 (rev 01) > 00:10.0 USB Controller: VIA Technologies, Inc. USB (rev 80) > 00:10.1 USB Controller: VIA Technologies, Inc. USB (rev 80) > 00:10.2 USB Controller: VIA Technologies, Inc. USB (rev 80) > 00:10.3 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 82) > 00:11.0 ISA bridge: VIA Technologies, Inc. VT8235 ISA Bridge > 00:11.1 IDE interface: VIA Technologies, Inc. VT82C586/B/686A/B PIPC Bus > Master IDE (rev 06) > 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233 AC97 > Audio Controller (rev 50) > 00:12.0 Ethernet controller: VIA Technologies, Inc. VT6102 [Rhine-II] (rev > 74) > > #interrupts: > >CPU0 > 0:5944488IO-APIC-edge timer > 1: 1351IO-APIC-edge keyboard > 2: 0 XT-PIC cascade > 8: 1IO-APIC-edge rtc > 9: 0 IO-APIC-level acpi > 14:1720179IO-APIC-edge ide0 > 15: 50IO-APIC-edge ide1 > 17: 354543 IO-APIC-level eth0 > 18: 8304 IO-APIC-level aic7xxx > 21: 0 IO-APIC-level usb-uhci, usb-uhci, usb-uhci, ehci-hcd > 22: 0 IO-APIC-level VIA8233 > 23:223 IO-APIC-level eth1 > NMI: 0 > LOC:5944174 > ERR: 0 > MIS: 0 > > ###zaptel.conf > > loadzone = nl > defaultzone = nl > span=1,1,3,ccs,ami > bchan=1-2 > dchan=3 > > ###zapata.conf > context=default > switchtype=euroisdn > pridialplan=unknown > overlapdial=yes > signalling=bri_cpe_ptmp > usecallerid=yes > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcall
Re: [Asterisk-Users] chan_sccp2 testers needed
Robert: I am starting today on contributing my information i gained to the asterisk-docs project including detailed information how various phones can be put into service :-) After i did that, i'll update the info on voip-info and on the sccp pages. --jan Robert Lawrence ([EMAIL PROTECTED]) wrote: > Jan: > > I am willing to test the latest version as I am very interested in some > features of my cisco 7960s that are not in the SIP firmware. The only > problem is that I am having trouble finding good examples on how to > configure the phone / chan_sccp2. I'm used to IAX2 and SIP. I have > never touched sccp2/skinny before, but I have been reading everything I > can find on it. voip-info wiki and the chan-sccp2 site is sparse in > configuration information. > > If you have some example configs or pointers to more information for a > 7960, I would be very greatful if you could pass them along. > > Thanks, > > Robert > > Jan Czmok wrote: > > >Dear Skinny/SCCP lovers :-) > > > >I've just completed & uploaded to the cvs the newest version with fixed > >redial key AND implementation of speed dials. please test extensively > >and report any bugs. i know that the display is not yet set correctly > >but the buttons are working as expected. > > > >Enjoy testing... > > > >--jan > > > >(*1) http://chan-sscp.sf.net > >(*2) yes, bugtracker is down at the moment, will fix this tomorrow > >morning (in about 8 hours) > > > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Jan Czmok, Network Engineering & Support, Global Access Telecomm, Inc. Ph.: +49 69 299896-35 - fax: +49 69 299896-40 - sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco ubr924
I belive the 924 will do either MGCP or H323. I will start working on it today. Even if it DOES do MGCP, it's not guarenteed that * will like it. Any of your previous work (config files) would be of assistance. I am going to start on it today. Thanks Duane Cox - Original Message - From: Gabriel Millerd To: [EMAIL PROTECTED] Sent: Friday, July 30, 2004 10:29 AM Subject: Re: [Asterisk-Users] cisco ubr924 Hey list, Does anyone have a current working config example of a ciscoubr924 and * ? I think the 924 only supports MGCP.I want to get VoIP on this device, I was wondering if anyone hasalready tackled the problem, if not, I'll go in blind :)i have tried myself (and posted) with not avail. If its possible tolearn from your progress please lemme know. If there is anything I canhelp with also let me know.I believe only H323 is an option for connectivity. I was neversuccessful in getting it to work with opengk and the like. However itsnot like that stuff is anywhere as friendly as * isThanks.
Re: [Asterisk-Users] Running AGI script on answer.
On Fri, 2004-07-30 at 06:22, David Wilson wrote: > Hi there, > > Is there an accepted way of running an AGI script on answering of a > channel? Is it even possible? I don't need to execute AGI commands, I > just need to know a channel has been answered. exten => 12345,1,Answer() exten => 12345,2,system(my_stupid_app.sh) exten => 12345,3,Goto(My_normal_behaviour,s,1) -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoring spool
On Fri, 2004-07-30 at 08:05, Lorenzo Fascì wrote: > Hi > > I have to send 4 outgoing call, all from ZAP1/1 when > asterisk start the first call, immediately it start all other calls > in the log I see > > Unable to request channel Zap/1/282 > Jul 30 15:05:27 NOTICE[-1242534992]: pbx_spool.c:232 attempt_thread: > Call failed to go through, reason 0 > > What I have to do to make asterisk make calls in sequence ? Allow enough retries so it will complete, or set the time in the future, or have call completion trigger the dropping of the next queue call in the queue, or get more hardware... Boy that simple. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing
Roman Bessyadovskii wrote:
Yes, I can make a call on that extension from other definity phone, if you
mean it.
-Original Message-
From: Ken Godee [mailto:[EMAIL PROTECTED]
Sent: 30 ÉÀÌÑ 2004 Ç. 19:14
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1.
Incoming work, but not outgoing
Roman Bessyadovskii wrote:
Hi All.
I connect asterisk and definity by manual at
www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya.
(I just only have E1, not T1 card).
I see, that card work (in definity trunk status, and at asterisk
Incoming call, from definity is work ok, but when I try outgoing call, I
recive
-- Executing Dial("SIP/1015-870f", "Zap/g1/2073") in new stack
-- Called g1/2073
-- Channel 1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
How fix it?
Do you have the "Dial Plan" set up properly
on the Definity side?
No, that's not what I mean.
You must understand how the "Dial plan" is used in the Definity.
Are all your extensions on the Definity 4 digits starting with a 2?
If you do not have first digit "2", length "4", type "extension"
set up in the dial plan, Definity will not know what to do with the four
digits your sending into the switch.
do a "display dialplan" and make sure it is set up correctly.
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Re: [Asterisk-Users] OH323 and codec selection
Chris A. Icide wrote: I'm having a small issue with the oh323 implementation when it comes to codec selection. Version info: CVS Head 6/30/2004 OH323 0.6.3 OpenPhone for windows version 1.8.1 Asterisk is configured as a h323 endpoint which either terminates to the PSTN locally through a PRI or terminates the h323 call to an IAX provider remotely. Asterisk also has G729 licences installed. in oh323.conf we set codecs allowed in the following order: G729 GSM ULAW ALAW When dialing in with OpenPhone with all codecs besides g729 disabled in the audio codec configuration panel, oh323 in Asterisk still picks and uses GSM as the selected codec. Only if I disable all but G729 in oh323.conf will Asterisk use G729 for an incomming h323 call. Am I doing something wrong? Is the order of the codecs in the oh323.conf significant, or is some other method of codec selection being used? Yes, the order of the codecs in oh323.conf is significant. I use it with ohphone (v1.4.3) without problems. Just make sure that you declare the codec of openphone as the preferred one. I don't know what happens when you just disable the codecs in openphone (and it seems that it doesn't work, since the codec that is selected is one of the disabled codecs). -Chris Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Limit incoming calls to SIP Channels
On July 30, 2004 03:40 pm, Daniel Niasoff wrote:
> Hi All,
>
> Can someone please tell me how to limit incoming calls to SIP channels
> using the SetGroup & Checkgroup command. I don't want any call waiting on
> SIP channels and you are somehow meant to be able to do it with these
> commands.
It's on the wiki @ http://voip-info.org/wiki-Asterisk+cmd+SetGroup
One thing I do is ensure I increment the the GROUPCOUNT for both the caller
and callee, to ensure the callee doesn't get a 2nd call while on the phone.
Do it something like:
[sip-phones]
; increment GROUPCOUNT for the calling exten
exten => 200,1,SetGroup(${CALLERIDNUM})
; increment GROUPCOUNT for exten you are calling
exten => 200,2,SetGroup(${EXTEN})
; ensure this is the 1st call to this exten
exten => 200,3,CheckGroup(1)
; dial if we make it this far
exten => 200,4,Dial(200)
; CheckGroup jumped here, 200 is on the phone
exten => 200,104,Busy
You will also want to ensure you SetGroup(${CALLERIDNUM}) before dialing any
outgoing numbers as well (to PSTN,etc). ie:
[outbound-local]
exten => _9NXX,1,SetGroup(${CALLERIDNUM})
Cheers
Ryan
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Re: [Asterisk-Users] zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0)
Hi, I have only just started watching this thread, so im not sure if you have posted it already, but if you have not already done so, please could you post the output from /proc/zaptel/ (i.e. /proc/zaptel/1) Im not sure if I will be able to help, but it might give the mailing list some extra information that might help dignose the problem. Regards, Michael East - Original Message - From: Vasyl Rublyov To: [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 8:33 PM Subject: Re: [Asterisk-Users] zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) I really upset from this forum... from last ten posts nothing came back reasonable. None for PRI problem, none for random disconnects Vasyl Rublyov wrote: Thank you. It works to me as well.. but some time I am getting (PRI side only) huge noise. reseaux wrote: Dear Vasyl I have a E100P on my Asterisk Box connect to Definity G3 and i run Asterisk CVS-HEAD-07/19/04-13:47:15 I dont see this kind of mex as you say but every minute i have this mex: Jul 28 15:26:40 NOTICE[229390]: chan_zap.c:7001 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 Jul 28 15:26:44 NOTICE[229390]: chan_zap.c:7001 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 Jul 28 15:27:28 NOTICE[229390]: chan_zap.c:7001 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 Jul 28 15:27:28 NOTICE[229390]: chan_zap.c:7001 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 --- But everything seems to work great in this box i dont have a very load use. Bye Dimitri On Wednesday 28 July 2004 01:41 pm, Vasyl Rublyov wrote: Anyone can comment this or just mailing is dead? Vasyl Rublyov wrote: I started to see this problem as soon as we connected to Verizon PRI (DMS-100 Switch) and it prints every 3-5 seconds. [Verizon DMS-100 PRI] <> [Lucent Merlin Legend] <> [Asterisk] Asterisk/LibPRI/Zaptel are built from HEAD CVS on Jul 10 2004. Any help? Jul 27 20:50:20 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) in pri debug: < Message type: ALERTING (1) < [97] < Locking Shift (len=01): Requested codeset 7 < [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:23 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! < Unknown IE 1857 (len = 16) Jul 27 20:30:23 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) < Protocol Discriminator: Unknown (0) len=22 < Call Ref: len= 2 (reference 0/0x0) (Originator) < Message type: ALERTING (1) < [97] < Locking Shift (len=01): Requested codeset 7 < [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:25 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! < Unknown IE 1857 (len = 16) Jul 27 20:30:25 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) < Protocol Discriminator: Unknown (0) len=22 < Call Ref: len= 2 (reference 0/0x0) (Originator) < Message type: ALERTING (1) < [97] < Locking Shift (len=01): Requested codeset 7 < [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:27 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! < Unknown IE 1857 (len = 16) Jul 27 20:30:27 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) < Protocol Discriminator: Unknown (0) len=22 < Call Ref: len= 2 (reference 0/0x0) (Originator) < Message type: ALERTING (1) < [97] < Locking Shift (len=01): Requested codeset 7 < [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:29 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! < Unknown IE 1857 (len = 16) Jul 27 20:30:29 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) /etc/zaptel.conf: # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us /etc/asterisk/zapata.conf: [channels] language=en context=default switchtype=national pridialplan=unknown overlapdial=no signalling=pri_net usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=yes callreturn=yes echocancel=32 echocancelwhenbridged=yes echotraining=yes rxgain=1.5 txgain=5.5 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no musiconhold=default channel => 1-23 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE
Re: [Asterisk-Users] Softphone - Freeware?!
> I assume this allows only one PreTransfer event at a time, i.e. things > will get messy if we have several users trying to PreTransfer their > calls at the same time? No. Each phone has its separate transfer environment. Here it's using the ParkAndAnnounce command which dynamically assigns a parking slot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue_log question: which endpoint was connected?
lenz wrote: Hello list, as I'm writing a little perl parser for queue_log analysis, I'd like to know *which* telephone answered a specific queue call. Unfortunately app_queue only logs the call id but does not log the call end point. This is okay for SIP endpoints, because their call id is something like SIP/endpointname-1234 so you can reasonably understand who was on answering, but for OH323 I get ID's like OH323/LJ5645 that are meaningless. Is there a way to extract from some other log the fact that OH323/LJ234 was a call placed to - say - OH323/[EMAIL PROTECTED] or can I extract it from some field of the peer data structure queue_log seems to extract data from? (to obtain call id, they gust print peer->name) The IP of the connected endpoint can be obtained from the OH323_RADDR variable. For incoming H.323 calls you can get the name of the channel and the IP address inside the dialplan, write them to a file and process them later. For outgoing H.323 calls [Dial(OH323/...)], you can't do it from the dialplan. In that case the OH323_RADDR variable is accessible only through the Dial() app. Anyway, it seems that the name of the OH323 channels needs to be more useful (added to my TODO list). Any help will be greatly appreciated. Thanks l. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit // incoming calls to Queue Agents
On July 30, 2004 09:19 am, Ryan Courtnage wrote: > On July 30, 2004 12:28 pm, Robert Jackson wrote: > > > I'm now struggling to find a way to similarily limit 2nd > > > calls to SIP clients > > > that are Agents, who receive their calls from a Queue(). Is > > > there any way to > > > accomplish this (without writing a patch)? > > > > I would also be very interested in this. My Agents also need to make > > outbound calls, but * tries to route calls to them even when they are on > > outbound calls. I ended up disabling call waitingfor those agents, and > > not allowing the calls to go to voicemail. > > I went ahead and submitted an enhancement request. Please review and add > to it if needed: > > http://bugs.digium.com/bug_view_page.php?bug_id=0002180 Scratch that - Mark already closed it :D I was using AddQueueMember() to add 'agents' to queues. I can get the functionality I require by using AgentCallbackLogin() instead. More details are in the 2180 bug notes. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Softphone - Freeware?!
> > What is the most complete Softphone application freeware? X-Lite is > very CooL, but the free version don´t support transfers... :( > Anyone know, a windows softphone free application that I can use > all > Askterisk Resources? Still a work in progress but quite effective (including transfers) IAX Phone is a good option. Check it out at: http://www.sokol-associates.com/ Thanks Steve Steven Sokol Owner/Manager Sokol & Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ASK ME ABOUT AstriCon 2004! http://www.astricon.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AstriCon 2004 Early Bird Period Ends Saturday!
Just a reminder that the 20% AstriCon 2004 Early Bird Discount expires this Saturday at midnight GMT. This translates to: 7:00 PM - Eastern Daylight Time 6:00 PM - Central Daylight Time 5:00 PM - Mountain Daylight Time 4:00 PM - Pacific Daylight Time If you want to save 20% off the conference fee, register now! Thanks, Steve & Olle Conference Organizers ASK ME ABOUT AstriCon 2004! http://www.astricon.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite to Asterisk through NAT?
Oh, hey. Sorry about that, I didn't see your reply in the plethora of Asterisk list posts. Hm...I'll try forwarding ports. I didn't try forwarding ports on the Windows box when I tried last. Greg Hill wrote: On Wed, 28 Jul 2004, programmer_ted wrote: I have an X-Lite phone on my box and I'm trying to register it with a remote Asterisk box. Both the X-Lite and Asterisk are behind a NAT. I know it's a pain to do because of SIP not working well with NATs, but I know there are ways to do such a thing...moving the Asterisk box outside the NAT is not a possibility at the moment. One thing we tried was mmm, a double-natted sip session. Now that's more fun than a person should be allowed to have in a single day. You didn't mention whether you have control over the NATs.. Everybody's favorite, port forwarding, may come to your rescue. It seems that x-lite always uses the same port for rtp (can't remember/find the number just now). Set the xp-side NAT to forward traffic on that port in to the xp box. You'd have to forward in the sip control port as well, I think. Then maybe do a similar thing on the * side (maybe you have to forward a large range of ports, 1-2 (?) on the * NAT?). I could be way off in the wrong ball field, though, so feel free to point out why this might not work. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 30 July 2004 09:30 am, Jozeph Brasil wrote: > Hi everybody, > > What is the most complete Softphone application freeware? X-Lite is > very CooL, but the free version don´t support transfers... :( > Anyone know, a windows softphone free application that I can use all > Askterisk Resources? > > Congratulations, > Jozeph Brasil. Hi Jozeph, Please do not use Reply to start a new thread as it starts in the middle of the thread you just replied to. Making it very hard to follow. Use New Message instead. You can right click on the address and choose New Message. - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBCobuljK16xgETzkRAt69AJ4g4JgadUALDEA86RdWwqjyh+EgYgCdF9l0 gMUKSzwWpaoZPT1bZ+cUuSQ= =vh6x -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite to Asterisk through NAT?
Did you have to forward any ports to the box running X-Lite (or is that one behind a NAT?) Thanks :) Ted Geoff Nordli wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of programmer_ted Sent: Wednesday, July 28, 2004 8:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X-Lite to Asterisk through NAT? Hi there, I have an X-Lite phone on my box and I'm trying to register it with a remote Asterisk box. Both the X-Lite and Asterisk are behind a NAT. I know it's a pain to do because of SIP not working well with NATs, but I know there are ways to do such a thing...moving the Asterisk box outside the NAT is not a possibility at the moment. One thing we tried was setting up a VPN, but I can't have the VPN server and a VPN client running on the same machine, because they use the same port (non-configurable). I can't set the VPN up on the Windows XP machine, because that only allows one user to be connected, and we need at least two. Any ideas? Thanks in advance, Ted Hi Ted. I managed to get it to work today. These are steps I took. On my firewall I port forwarded 5060, 1-11000 UDP to the internal Asterisk box. In the sip.conf file I made these changes: nat=yes externip = public.ip.address On the X-lite phone I pointed the SIP Proxy to the public.ip.address that was set above in the sip.conf file. Good Luck. Geoff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Two different voicemail messages
Hi all, I already have a voicemail.conf message defined... for one application... (users) "hi user, you have voice mail" Is it possible to have another voicemail message for another users ? (administrator) "hi adm, you have mail from 1234, etc etc.. more information" How can I do that ? Thanks a lot. Andrei. _ MSN Hotmail, o maior webmail do Brasil. http://www.hotmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X-Lite to Asterisk through NAT?
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of programmer_ted Sent: Friday, July 30, 2004 10:34 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X-Lite to Asterisk through NAT? Did you have to forward any ports to the box running X-Lite (or is that one behind a NAT?) Thanks :) Ted Geoff Nordli wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of programmer_ted Sent: Wednesday, July 28, 2004 8:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X-Lite to Asterisk through NAT? Hi there, I have an X-Lite phone on my box and I'm trying to register it with a remote Asterisk box. Both the X-Lite and Asterisk are behind a NAT. I know it's a pain to do because of SIP not working well with NATs, but I know there are ways to do such a thing...moving the Asterisk box outside the NAT is not a possibility at the moment. One thing we tried was setting up a VPN, but I can't have the VPN server and a VPN client running on the same machine, because they use the same port (non-configurable). I can't set the VPN up on the Windows XP machine, because that only allows one user to be connected, and we need at least two. Any ideas? Thanks in advance, Ted Hi Ted. I managed to get it to work today. These are steps I took. On my firewall I port forwarded 5060, 1-11000 UDP to the internal Asterisk box. In the sip.conf file I made these changes: nat=yes externip = public.ip.address On the X-lite phone I pointed the SIP Proxy to the public.ip.address that was set above in the sip.conf file. Good Luck. Geoff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I am sorry Ted, but I forgot about receiving calls on the X-lite client. This solution worked for outbound calls, but I don't think this solution will work for calls that are made to the x-lite client that is hiding behind the NAT device. The reason why outbound calls will work is because the NAT device was stateful so it will automatically keep the channel open. When calls are made from Asterisk to the X-lite device it will try to connect a given port but it won't be able to unless the port is open. It looks like there are some options within X-lite that allow you to nail down the RTP port and you could probably do some port forwarding on the NAT device based on the RTP port that you select. Does anyone have any recommendations on the configuration? Geoff --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date: 7/29/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New to IP-PBX
Hi, I'd really appreciate it if you can explain this to me. I have a D/41JCT-LS Dialogic board and I want to use it as an IP-PBX. I'm new to IP Telephony and telephony and general and I researched a lot but still confused about what I really need. I know that I can setup an IP-Telephony for my LAN using a SIP server and SIP compatible software phones. But the challenge is how can I connect to the PSTN so that I can send and receive calls? I came through a lot of terms like VoIP gateway and stuff like that but still don't know what I really need. Can you help? Thanks in advance Duraid ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Rodopi Billing
Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0)
Thank you. Here it is: $ cat /proc/zaptel/1 Span 1: WCT1/0 "Digium Wildcard T100P T1/PRI Card 0" B8ZS/ESF ClockSource 1 WCT1/0/1 ClearChannel (In use) 2 WCT1/0/2 ClearChannel (In use) 3 WCT1/0/3 ClearChannel (In use) 4 WCT1/0/4 ClearChannel (In use) 5 WCT1/0/5 ClearChannel (In use) 6 WCT1/0/6 ClearChannel (In use) 7 WCT1/0/7 ClearChannel (In use) 8 WCT1/0/8 ClearChannel (In use) 9 WCT1/0/9 ClearChannel (In use) 10 WCT1/0/10 ClearChannel (In use) 11 WCT1/0/11 ClearChannel (In use) 12 WCT1/0/12 ClearChannel (In use) 13 WCT1/0/13 ClearChannel (In use) 14 WCT1/0/14 ClearChannel (In use) 15 WCT1/0/15 ClearChannel (In use) 16 WCT1/0/16 ClearChannel (In use) 17 WCT1/0/17 ClearChannel (In use) 18 WCT1/0/18 ClearChannel (In use) 19 WCT1/0/19 ClearChannel (In use) 20 WCT1/0/20 ClearChannel (In use) 21 WCT1/0/21 ClearChannel (In use) 22 WCT1/0/22 ClearChannel (In use) 23 WCT1/0/23 ClearChannel (In use) 24 WCT1/0/24 HDLCFCS (In use) Michael wrote: Hi, I have only just started watching this thread, so im not sure if you have posted it already, but if you have not already done so, please could you post the output from /proc/zaptel/ (i.e. /proc/zaptel/1) Im not sure if I will be able to help, but it might give the mailing list some extra information that might help dignose the problem. Regards, Michael East - Original Message - From: Vasyl Rublyov To: [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 8:33 PM Subject: Re: [Asterisk-Users] zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) I really upset from this forum... from last ten posts nothing came back reasonable. None for PRI problem, none for random disconnects Vasyl Rublyov wrote: Thank you. It works to me as well.. but some time I am getting (PRI side only) huge noise. reseaux wrote: Dear Vasyl I have a E100P on my Asterisk Box connect to Definity G3 and i run Asterisk CVS-HEAD-07/19/04-13:47:15 I dont see this kind of mex as you say but every minute i have this mex: Jul 28 15:26:40 NOTICE[229390]: chan_zap.c:7001 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 Jul 28 15:26:44 NOTICE[229390]: chan_zap.c:7001 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 Jul 28 15:27:28 NOTICE[229390]: chan_zap.c:7001 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 Jul 28 15:27:28 NOTICE[229390]: chan_zap.c:7001 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 --- But everything seems to work great in this box i dont have a very load use. Bye Dimitri On Wednesday 28 July 2004 01:41 pm, Vasyl Rublyov wrote: Anyone can comment this or just mailing is dead? Vasyl Rublyov wrote: I started to see this problem as soon as we connected to Verizon PRI (DMS-100 Switch) and it prints every 3-5 seconds. [Verizon DMS-100 PRI] <> [Lucent Merlin Legend] <> [Asterisk] Asterisk/LibPRI/Zaptel are built from HEAD CVS on Jul 10 2004. Any help? Jul 27 20:50:20 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) in pri debug: < Message type: ALERTING (1) < [97] < Locking Shift (len=01): Requested codeset 7 < [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:23 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! < Unknown IE 1857 (len = 16) Jul 27 20:30:23 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) < Protocol Discriminator: Unknown (0) len=22 < Call Ref: len= 2 (reference 0/0x0) (Originator) < Message type: ALERTING (1) < [97] < Locking Shift (len=01): Requested codeset 7 < [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:25 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! < Unknown IE 1857 (len = 16) Jul 27 20:30:25 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) < Protocol Discriminator: Unknown (0) len=22 < Call Ref: len= 2 (reference 0/0x0) (Originator) < Message type: ALERTING (1) < [97] < Locking Shift (len=01): Requested codeset 7 < [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:27 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! < Unknown IE 1857 (len = 16) Jul 27 20:30:27 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) < Protocol Discriminator: Unknown (0) len=22 < Call Ref: len= 2 (reference 0/0x0) (Originator) < Message type: ALERTING (1) < [97] < Locking Shift (len=01): Requested codeset 7 < [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:29 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! < Unknown IE 1857 (len = 16) Jul 27 20:30:29 WARNING[98310]: chan_zap.c:6665 zt_pri_erro
[Asterisk-Users] Asterisk behind openBSD firewall/NAT
Hi All: Has anybody been able to get Asterisk work behind a openBSD firewall/NAT? If you have then would it be possible to share your pf.config file? I am trying to get Asterisk which is behind an openBSD firewall/NAT to register with FWD but can't get it to talk. I have captured IP traffic on udp port 5060 using tcpdump on both internal and external interfaces of my openBSD gateway (the logs are at the end of this e-mail). From the tcpdump logs I can see that a message is sent to FWD out from the external interface and response is received from FWD on udp port 5060 but the response is not forwarded to Asterisk. Here are my NAT and FILTER rules: (tl0 is the external interface, xl0 is the internal interface) nat on tl0 inet from 192.168.0.0/24 to any -> (tl0) round-robin rdr on xl0 inet proto tcp from any to any port = ftp -> 127.0.0.1 port 8021 rdr pass on tl0 inet proto tcp from any to (tl0) port = sip -> 192.168.0.3 rdr pass on tl0 inet proto udp from any to 209.89.66.243 port = sip -> 192.168.0.3 port 5060 Tcpdump output on the external interface of gateway (filtered for udp port 5060): tcpdump: listening on tl0 05:51:58.622714 243.209-89-66-0.interbaun.com.57199 > 192.246.69.223.sip: udp 376 (DF) (ttl 63, id 23576) 05:51:58.716031 192.246.69.223.sip > 243.209-89-66-0.interbaun.com.sip: udp 462 (DF) (ttl 47, id 0) 05:51:59.622771 243.209-89-66-0.interbaun.com.57199 > 192.246.69.223.sip: udp 376 (DF) (ttl 63, id 23577) 05:51:59.716004 192.246.69.223.sip > 243.209-89-66-0.interbaun.com.sip: udp 462 (DF) (ttl 47, id 0) 05:52:00.623539 243.209-89-66-0.interbaun.com.57199 > 192.246.69.223.sip: udp 376 (DF) (ttl 63, id 23578) 05:52:00.719989 192.246.69.223.sip > 243.209-89-66-0.interbaun.com.sip: udp 462 (DF) (ttl 47, id 0) 05:52:01.624328 243.209-89-66-0.interbaun.com.57199 > 192.246.69.223.sip: udp 376 (DF) (ttl 63, id 23579) 05:52:01.716980 192.246.69.223.sip > 243.209-89-66-0.interbaun.com.sip: udp 462 (DF) (ttl 47, id 0) 05:52:02.624107 243.209-89-66-0.interbaun.com.57199 > 192.246.69.223.sip: udp 376 (DF) (ttl 63, id 23580) 05:52:02.715968 192.246.69.223.sip > 243.209-89-66-0.interbaun.com.sip: udp 462 (DF) (ttl 47, id 0) 05:52:03.623884 243.209-89-66-0.interbaun.com.57199 > 192.246.69.223.sip: udp 376 (DF) (ttl 63, id 23581) 05:52:03.715954 192.246.69.223.sip > 243.209-89-66-0.interbaun.com.sip: udp 462 (DF) (ttl 47, id 0) 05:52:18.645348 243.209-89-66-0.interbaun.com.57199 > 192.246.69.223.sip: udp 376 (DF) (ttl 63, id 23582) 05:52:18.737143 192.246.69.223.sip > 243.209-89-66-0.interbaun.com.sip: udp 462 (DF) (ttl 47, id 0) Tcpdump output at the internal interface tcpdump: listening on xl0 06:05:00.451172 192.168.0.3.sip > fwd.pulver.com.sip: udp 376 (DF) (ttl 64, id 23811) 06:05:01.450934 192.168.0.3.sip > fwd.pulver.com.sip: udp 376 (DF) (ttl 64, id 23812) 06:05:02.450711 192.168.0.3.sip > fwd.pulver.com.sip: udp 376 (DF) (ttl 64, id 23813) 06:05:03.451502 192.168.0.3.sip > fwd.pulver.com.sip: udp 376 (DF) (ttl 64, id 23814) 06:05:04.451286 192.168.0.3.sip > fwd.pulver.com.sip: udp 376 (DF) (ttl 64, id 23815) Regards, Karim Mardhani ZeeCore Consulting ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to IP-PBX
you need www.voip-info.org - Original Message - From: "Duraid Abbas" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 2:39 PM Subject: [Asterisk-Users] New to IP-PBX Hi, I'd really appreciate it if you can explain this to me. I have a D/41JCT-LS Dialogic board and I want to use it as an IP-PBX. I'm new to IP Telephony and telephony and general and I researched a lot but still confused about what I really need. I know that I can setup an IP-Telephony for my LAN using a SIP server and SIP compatible software phones. But the challenge is how can I connect to the PSTN so that I can send and receive calls? I came through a lot of terms like VoIP gateway and stuff like that but still don't know what I really need. Can you help? Thanks in advance Duraid ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer call help needed
I have * installed as a voicemail only solution attached to a Toshiba Strata phone system. I am testing * at the current time, so it is not attached to ports designated as VM on the Toshiba. I am not sure if the VM ports would handle the call any different than an analog port. Currently if I call one of the DIDs that I have connected to the *, my greeting plays and I am able to dial my extension. The problem with my current setup is I use two ports for every call that comes in. One port is used for the inbound call and one port is used for calling the extension. I have looked over the 'Transfer' command, but I am unsure of the syntax. All my phone lines come in to the Toshiba on pots lines. I have one tdm04b installed in my * system. Once I can accomplish my goal, I will order an additional tdm04b. Here is a breakdown of how I want a call to be handled: 1. Person calls company 2. The Toshiba hands the call to * via automated attendant 3. The caller is asked to enter the extension of the person he/she is trying to reach. 4. The call is transferred to the extension. If there is no answer at the extension entered, the call is sent to the voicemail for the extension. If I use the 'Transfer' command, will the caller be put in to voicemail if there is no answer? Also, does the 'Transfer' command use only one port or does it require two ports? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323-0.6.3a
Hi there. I thy to compile asterisk-oh323-0.6.3a but it fail in the make command. I have the pwlib-v1_6_6-1 and openh323-v1_13_5-1 as saying in the README file of the packet asterisk-oh323-0.6.3a I do make and this is the error: # make for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make: *** No rule to make target `ccflags'. Stop. make: *** No rule to make target `ccflags'. Stop. make[1]: Entering directory `/root/playmonkeys/asterisk-oh323-0.6.3a/wrapper' ./check_ver /home/admin/h323/pwlib pwlib ./check_ver /home/admin/h323/openh323 openh323 g++ -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\"1.6.6\" -DOPENH 323VERSION=\"1.13.5\" -I/home/admin/h323/pwlib/include/ptlib/unix -I/home/a dmin/h323/pwlib/include -I/home/admin/h323/openh323/include -I/home/admin/h3 23/openh323/include/openh323 -I../asterisk-driver -c wrapper_misc.cxx -o wrapper_misc.o In file included from /home/admin/h323/pwlib/include/ptlib.h:169, from wrapper_misc.hxx:35, from wrapper_misc.cxx:34: /home/admin/h323/pwlib/include/ptlib/unix/ptlib/pdirect.h:78: error: syntax error before `protected' /home/admin/h323/pwlib/include/ptlib/unix/ptlib/pdirect.h:80: error: syntax error before `*' token In file included from /home/admin/h323/pwlib/include/ptlib.h:181, from wrapper_misc.hxx:35, from wrapper_misc.cxx:34: /home/admin/h323/pwlib/include/ptlib/unix/ptlib/config.h:53: error: syntax error before `public' /home/admin/h323/pwlib/include/ptlib/unix/ptlib/config.h:55: error: destructors must be member functions /home/admin/h323/pwlib/include/ptlib/unix/ptlib/config.h:57: error: syntax error before `protected' . continue thanks in advance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Non standard usage of X100P card.
If it were "rather simple", this would have been done a long time ago. Signaling requirements for FXS and FXO are considerably different, however, so you can't simply abuse a modem to drive a phone. Consider the IAXy or the Digium Dev-Kit, which includes a USB/FXS adapter. > -Original Message- > From: Dmitry Sergeev [mailto:[EMAIL PROTECTED] > Sent: Friday, July 30, 2004 4:18 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Non standard usage of X100P card. > > > I have two X100P card in my box. I want to connect regular > phone (not the > phone line!) to one of thse cards. Does anybody think about the same? > > I don't really want an expensive solution buying additional > card with FXS > port, I prefer to make something by myself. It'll be great if > somebody can > point me to technical materials or show electric scheme of > such converter. I > believe it should be rather simple. > > I'm ready to cooperate to make such solution. > > Thanks, > Dmitry. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New to IP-PBX
To connect to the PSTN (existing line), you'll need an FXO port. Easiest way to get one is to put in a Digium X100P card. Not sure whether the Dialogic board is compatible with Asterisk, nor even what it does. > -Original Message- > From: Duraid Abbas [mailto:[EMAIL PROTECTED] > Sent: Friday, July 30, 2004 1:39 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] New to IP-PBX > > > Hi, > > I'd really appreciate it if you can explain this to me. > > I have a D/41JCT-LS Dialogic board and I want to use it as an > IP-PBX. I'm new to IP Telephony and telephony and general and > I researched a lot but still confused about what I really need. > > I know that I can setup an IP-Telephony for my LAN using a > SIP server and SIP compatible software phones. But the > challenge is how can I connect to the PSTN so that I can send > and receive calls? > > I came through a lot of terms like VoIP gateway and stuff > like that but still don't know what I really need. Can you help? > > Thanks in advance > > Duraid ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outgoing *-initiated calls from spool directory not working
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 my test.call file (my cellphone and did have been censored for privacy): gf-002-pbx-001 outgoing # cat test.call Channel: Zap/g1/1313529 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: dialplan Extension: 1248xxx Priority: 1 Callerid: 313529 gf-002-pbx-001 outgoing # i have that context/extension/priority in my dialplan, but i don't think it's even getting that far. placing a new file in /var/spool/asterisk/outgoing causes no change in /var/log/asterisk/messages, nor on the console. it's as if it's not even checking for new files. Regards, - -j - -- Jeffrey Paul, Senior Network Administrator - [EMAIL PROTECTED] Group Financial LLC / Diamond Financial Products 4000 Town Center/Suite 1000/Office 1013/Southfield/MI/48075-1501 o: 800-476-5882 x244 m: 800-476-5882 x468 f: 800-510-1405 DE2B 3F61 14A1 BD0F B496 DC91 3D97 8C4D 7678 4A42 - -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Stingel Sent: Friday, July 30, 2004 11:20 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Outgoing *-initiated calls from spool directory not working Hi Jeffrey- I use the outgoing call feature all the time and it seems to work ok, but haven't updated CVS in a couple weeks, so maybe there's a new bug. Just for fun, please post the contents of one of your call files. Also, just checking that you find nothing in /var/log/asterisk/messages? Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California & London England www.evtmedia.com - -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeffrey Paul Sent: Friday, July 30, 2004 8:05 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Outgoing *-initiated calls from spool directory not working I'm running: Asterisk CVS-HEAD-07/06/04-17:49:49 built by [EMAIL PROTECTED] on a i686 running Linux I've tried placing files (both ending in .call and not) in the correct format in /var/spool/asterisk/outgoing I get _nothing_. No log messages, nothing on the console, zip. Permissions seem to be correct on both the files and the directory, as well (* is running as root, for right now during testing). gf-002-pbx-001 root # grep astspooldir /etc/asterisk/asterisk.conf astspooldir => /var/spool/asterisk gf-002-pbx-001 root # grep ^console /etc/asterisk/logger.conf console => notice,warning,error,debug gf-002-pbx-001 root # ls -tla /var/spool/asterisk/outgoing total 12 drwxrwxrwx 2 root root 4096 Jul 30 15:00 . - -rw-r--r-- 1 root root 1292 Jul 26 05:08 test.call drwxr-xr-x 5 root root 4096 Jul 25 23:14 .. I've tried mv'ing the files there, tried cp'ing the files there, tried creating them in place... Nothing. I've tried looking around for documentation on this issue, but I just can't find where I'm going wrong. Any suggestions are appreciated. Thanks in advance, - -j - -- Jeffrey Paul, Senior Network Administrator - [EMAIL PROTECTED] Group Financial LLC / Diamond Financial Products 4000 Town Center/Suite 1000/Office 1013/Southfield/MI/48075-1501 o: 800-476-5882 x244 m: 800-476-5882 x468 f: 800-510-1405 DE2B 3F61 14A1 BD0F B496 DC91 3D97 8C4D 7678 4A42 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: PGP 8.0.2 iQA/AwUBQQquZT2XjE12eEpCEQJaQwCgkWbXmMq+XielyrIjVKdNPHnIO04An129 s0h75SjCuygh3YCCVPthG1E2 =hr2D -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to IP-PBX
Hello, On Fri, 2004-07-30 at 15:39, Duraid Abbas wrote: > I have a D/41JCT-LS Dialogic board and I want to use it as an IP-PBX. > I'm new to IP Telephony and telephony and general and I researched a lot > but still confused about what I really need. > > I know that I can setup an IP-Telephony for my LAN using a SIP server > and SIP compatible software phones. But the challenge is how can I > connect to the PSTN so that I can send and receive calls? Asterisk will do a wonderfull job as a soft PBX, but my advice is to use hardware from Digium to connet to the PSTN (FXO or T1/E1) and to connect regular analog phones (FXS or T1/E1+ChannelBank): http://www.digium.com/index.php?menu=hardware_products Before purchasing hardware, you can try to set up Asterisk just with SIP softphones and get it to know the platform. Once you are comfortable you can jump on buying some hardware. If you do not have time to investigate yourself search for "Asterisk consultants" on http://www.voip-info.org Best regards, -- Nicolas Gudino <[EMAIL PROTECTED]> House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Conference Call?
On Thu, 29 Jul 2004, Mike Machado wrote: > > > Another thing you could do is use a regular phone to call into a DID and > > enter the conference, then everybody can join that conference and listen. No > > bandwidth required, just a phone call to the distributor's Asterisk server. > > Then just keep that phone near the person speaking, like a microphone. > > > > I might be able to donate some dial in lines for this. I have local > numbers for California, but outside would be a long distance call. I can > probably get MP3 streaming of the conference working too. > > Whoever is organizing the conference can contact me off list if they are > interested and don't already have another solution. I can donate a conference room, and a PRI's worth of Dial-Up in the 216 and 440 Area codes (Cleveland, Ohio). -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to IP-PBX
The Dialogic boards are intel's version of the wildcard adaptors... I believe the one he's referring to has like 4 FXO Ports, just like the TDM400P... I think I've read that dialogic boards *ARE* compatible with *, but have not seen any specific examples of such a configuration... - Original Message - From: "Jay Milk" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 1:23 PM Subject: RE: [Asterisk-Users] New to IP-PBX > To connect to the PSTN (existing line), you'll need an FXO port. > Easiest way to get one is to put in a Digium X100P card. Not sure > whether the Dialogic board is compatible with Asterisk, nor even what it > does. > > > -Original Message- > > From: Duraid Abbas [mailto:[EMAIL PROTECTED] > > Sent: Friday, July 30, 2004 1:39 PM > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] New to IP-PBX > > > > > > Hi, > > > > I'd really appreciate it if you can explain this to me. > > > > I have a D/41JCT-LS Dialogic board and I want to use it as an > > IP-PBX. I'm new to IP Telephony and telephony and general and > > I researched a lot but still confused about what I really need. > > > > I know that I can setup an IP-Telephony for my LAN using a > > SIP server and SIP compatible software phones. But the > > challenge is how can I connect to the PSTN so that I can send > > and receive calls? > > > > I came through a lot of terms like VoIP gateway and stuff > > like that but still don't know what I really need. Can you help? > > > > Thanks in advance > > > > Duraid > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Astricon Conference Call?
On Thu, 29 Jul 2004, Steve Woolley wrote: > I thought the interesting exercise would be to use asterisk for the > task. Couldn't we use a kind of distributed conference call where a few > key select/high bandwidth asterisk servers form the main conference and > then have multiple layers of conferencing. That way no one > server/network is saturated. As long as you ensure that no-one can SPEAK in the conference, this should be just fine. The Conference could be beamed out to a single Asterisk server (The Master) and then distributed via IAX2 and MeetMe to a bunch of servers in different area codes (The Slaves). We do this all the time between my IAX2/[EMAIL PROTECTED]/4570 bridge, BKW's system and occasionally the NuFone MeetMe. It would be trivial to coordinate. I can provide a high-bandwidth distribution point, and NuFone has offered the same. I don't see any reason why this can't be easily accomplished. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP connections do not hang up
Hi everybody, I have strange problem: I'm calling from inside (either X-Lite using SIP channel or a ISDN telephone using Zap Channel) using sipgate to a number in public network. When I'm hanging up before the other person picked up the phone, the line is not closed correctly. The phone keeps on ringing until timeout (of Sipgate I assume) and it even costs my money, if the other person picks up the ringing phone, even if I already hung up. The register entries come when I reject the call on the public network phone. I hope anyone can help me! Thanx in advance, Florian PS: Hanging up when using CAPI instead of SIP works perfectly... sip.conf: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = psin ; Default context for incoming calls tos=lowdelay; IP QoS parameter, either keyword or value maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration disallow=all; Disallow all codecs allow=alaw allow=ulaw ; Allow codecs in order of preference allow=g726 allow=gsm allow=ilbc musicclass=random externip = rauserv.dyndns.org ; Address that we're going to put in outbound SIP messages localnet=192.168.0.0/255.255.255.0; All RFC 1918 addresses are local networks language=de register => 888:[EMAIL PROTECTED]/2001 [sipgate] type = friend username = 888 canreinvite=no secret = aaa host = sipgate.de context=psin fromuser = 888 fromdomain = sipgate.de nat = no qualify = yes insecure=very ;pickupgroup=1 ;callgroup=1 ;dtmfmode=rfc2833 == See sip debug output: To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 30 Jul 2004 20:40:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 314 v=0 o=root 32131 32131 IN IP4 82.83.56.91 s=session c=IN IP4 82.83.56.91 t=0 0 m=audio 15080 RTP/AVP 8 0 2 3 97 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 217.10.79.9:5060 -- Called [EMAIL PROTECTED] Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 2 Found RTP audio format 101 Peer RTP is at port 217.10.64.78:0 Found description format PCMA Found description format PCMU Found description format GSM Found description format iLBC Found description format G726-32 Found description format telephone-event Capabilities: us - 0x41e(GSM|ULAW|ALAW|G726|ILBC), peer - audio=0x41e(GSM|ULAW|ALAW|G726|ILBC)/video=0x0(EMPTY), combined - 0x41e(GSM|ULAW|ALAW|G726|ILBC) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) -- SIP/sipgate-76e5 is making progress passing it to SIP/2112-495b Reliably Transmitting: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK0b9e12cc From: "Florian" ;tag=as1c3c2263 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 217.10.79.9:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms == Spawn extension (out, 907141220856, 3) exited non-zero on 'SIP/2112-495b' Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK0b9e12cc From: "Florian" ;tag=as1c3c2263 To: ;tag=cbf5cb1d0d4e31526039b4f3671ccf51-3d18 Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 217.10.79.9:5060 Destroying call '[EMAIL PROTECTED]' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:217.10.79.9 SIP/2.0 Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK2eefa4de From: "asterisk" ;tag=as7c4322e5 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Fri, 30 Jul 2004 20:41:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 217.10.79.9:5060 Destroying call '[EMAIL PROTECTED]' 11 headers, 0 lines Reliably Transmitting: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK388496a5 From: ;tag=as7cc829e2 To: Call-ID: [EMAIL PROTECTED] CSeq: 108 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.10.79.9:5060 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK0b70e7c4 From: ;tag=as7cc829e2 To: Call-ID: [EMAIL PROTECTED] CSeq: 109 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="888", realm="sipgate.de", algorithm=MD5, uri="sip:sipgate.de", nonce="410ab3a95b427f0b044db27cdada37d268edcfee", response="f4b21e167ad3b177beddc
[Asterisk-Users] FIREFLY repeat calls
Hi! I´m trying to use firefly 3 party with * and iax2. I cant figure out why it reapeats every call many times until it is closed. It is a bug ? I want it because of the skin changing thing.. Someone have a clue on how to use it with * Thanks Miklos - Original Message - From: "Jozeph Brasil" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 11:17 AM Subject: RES: [Asterisk-Users] Softphone - Freeware?! I have one X100P installed with two SIP extensions using X-Lite, I just would like to transfer the call to another SIP extension; Just a "Flash"+"Extension"+"Hangup CALL"... Thanks for all! -Mensagem original- De: Eric Bart [mailto:[EMAIL PROTECTED] Enviada em: sexta-feira, 30 de julho de 2004 10:51 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Softphone - Freeware?! axra will do. it's an add-on that will give consultative transfer to X-Lite (and others). see below : --- New application for asterisk : axra axra runs separately. developped in C++. it dialogs with asterisk through agi calls and through the manager api. it proccesses phone calls through the dial plan (agi) and concurently through the manager api. axra currently provides consultative transfer for SIP and IAX2 phones. this should easily be extended to any phone technology. hopefully, axra will soon provide 3 way calling. there are two tranfer functions : PreTransfer and CTransfer each should be implemented in the dial plan like : exten => 76,1,AGI(axraagi|PreTransfer) exten => 76,2,Hangup exten => 77,1,AGI(axraagi|CTransfer|auto) exten => 77,2,Hangup you may choose other extensions than 76 & 77. you may omit 'auto' when a call is transfered to PreTransfer (76), the call is parked and waits for a transfer. if the timeout occurs, the call is ringed back. if you call PreTransfer (76) directly, the parked call (if any) is immediatly ringed back. when a call is transfered to CTransfer (77), the call is linked to the pretransfered (parked) call. if no pretransfer exists the call is pretransfered just like 76 was dialed. however, if 'auto' was specified, axra will try to link the call to the oldest live channel attached to transferer's phone. http://www.byortek.com/asterisk/axra-2004-07-29.tgz Please download, read REAME and INSTALL. Any feedback greatly appreciated. - Original Message - From: "Jozeph Brasil" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 3:30 PM Subject: [Asterisk-Users] Softphone - Freeware?! > Hi everybody, > > What is the most complete Softphone application freeware? X-Lite is > very CooL, but the free version don´t support transfers... :( > Anyone know, a windows softphone free application that I can use all > Askterisk Resources? > > Congratulations, > Jozeph Brasil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to IP-PBX
I have been seeing reccomendations for using asterisk as a soft-pbx with the reccomendation being to use regular analog phones via FXS rather than SIP. Is this still a big issue? Or is this a left-over from previous bad experiences? I have been doing demos with SIP phones, and some IAXYs to whet their apetites, and people are really biting at the feature set I can provide, and I have run into no problems yet, but I would love to know at what threshold of SIP phones does the system start to have problems. The assumption in my scenario is a quality ASUS motherboard, running RedHat/Debian, 512 MB RAM 10/100 Ethernet, P4 2.4 Ghz processor. I am trying to hit the small office market, with up to 20 SIP phones, and up to 8 POTS lines. (These have been my current limits until I see the system inaction a bit more) Is the problem in using dissimilar SIP phones with different codecs? Thus burdening the processor with conversion on top of all of the other work it is doing? PS, I am having a whale of a time with this software, and I appreciate the helpfullness of members of the community... Jim Richards Kissyfish On Fri, 2004-07-30 at 15:31, Nicolas Gudino wrote: > Hello, > > On Fri, 2004-07-30 at 15:39, Duraid Abbas wrote: > > I have a D/41JCT-LS Dialogic board and I want to use it as an IP-PBX. > > I'm new to IP Telephony and telephony and general and I researched a lot > > but still confused about what I really need. > > > > I know that I can setup an IP-Telephony for my LAN using a SIP server > > and SIP compatible software phones. But the challenge is how can I > > connect to the PSTN so that I can send and receive calls? > > Asterisk will do a wonderfull job as a soft PBX, but my advice is to use > hardware from Digium to connet to the PSTN (FXO or T1/E1) and to connect > regular analog phones (FXS or T1/E1+ChannelBank): > > http://www.digium.com/index.php?menu=hardware_products > > Before purchasing hardware, you can try to set up Asterisk just with SIP > softphones and get it to know the platform. Once you are comfortable you > can jump on buying some hardware. > > If you do not have time to investigate yourself search for "Asterisk > consultants" on http://www.voip-info.org > > Best regards, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Dev Meeting On Line
The idea is to have people attend the conference and not primarily a webcast. Granted due to the nature of the community it's not possible for everyone that would like to attend to justify it for cost reasons and distance. I think the Dev meeting is best fitting to be broadcasted in this manner anyway due to the number of presentations going on at once during the other days of the conference. Just my thoughts on it =) On Fri, 30 Jul 2004 09:39:06 -0400, Steve Woolley <[EMAIL PROTECTED]> wrote: > > Only the Developer's Meeting will be considered for broadcast > > at this time. > > Why? > > Seems there are a large number of individuals willing to donate > bandwidth and CPU cycles for this. > > -- > Steve Woolley > IT Manager > ADS Telecom, Inc. > 59 Skyline Drive > Suite 1250 > Lake Mary, Florida 32746 > > Phone: (407)682-6226 x1110 > Fax: (407)682-3455 > Cell: (321)229-5311 > > [EMAIL PROTECTED] > www.adstelecom.com > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP connections do not hang up
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 If you just bothered to search this list in the past 12 hours, you would have found a solution around that: to summarize: Add in zapata.conf: busydetect=yes busycount=6 The maximum it will take for asterisk to see the person hanged-up is after 6 busy dial-tones. On 31/07/2004, at 6:58 AM, Florian Rau wrote: I'm calling from inside (either X-Lite using SIP channel or a ISDN telephone using Zap Channel) using sipgate to a number in public network. When I'm hanging up before the other person picked up the phone, the line is not closed correctly. The phone keeps on ringing until timeout (of Sipgate I assume) and it even costs my money, if the other person picks up the ringing phone, even if I already hung up. - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBCsHLXeDVKqIr3GURArjyAJ9p97F/wWIiIesaYo85QfHut8zbzQCgj2l2 uuKZxyJoaSmpI9V9I+ojnJc= =Y8jQ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New to IP-PBX
If you don't do any transcoding, and turn canreinvite=on for your sip-clients, there shouldn't be a reason why you couldn't run hundreds or thousands of extensions on a Celeron 500. Once you get into transcoding (or you turn canreinvite=off in order to allow for recording of conversations), processor speed matters. AFAIK, the #1 reason for recommending POTS over SIP is that in an all-IP system, you'll need a timing source, and that can be tricky on some systems. > -Original Message- > From: James Richards [mailto:[EMAIL PROTECTED] > Sent: Friday, July 30, 2004 4:18 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] New to IP-PBX > > > I have been seeing reccomendations for using asterisk as a > soft-pbx with the reccomendation being to use regular analog > phones via FXS rather than SIP. > > Is this still a big issue? Or is this a left-over from > previous bad experiences? I have been doing demos with SIP > phones, and some IAXYs to whet their apetites, and people are > really biting at the feature set I can provide, and I have > run into no problems yet, but I would love to know at what > threshold of SIP phones does the system start to have problems. > > The assumption in my scenario is a quality ASUS > motherboard, running RedHat/Debian, 512 MB RAM 10/100 > Ethernet, P4 2.4 Ghz processor. > > I am trying to hit the small office market, with up to 20 > SIP phones, and up to 8 POTS lines. (These have been my > current limits until I see the system inaction a bit more) > > Is the problem in using dissimilar SIP phones with > different codecs? Thus burdening the processor with > conversion on top of all of the other work it is doing? > > PS, I am having a whale of a time with this software, and I > appreciate the helpfullness of members of the community... > > Jim Richards > Kissyfish > > On Fri, 2004-07-30 at 15:31, Nicolas Gudino wrote: > > Hello, > > > > On Fri, 2004-07-30 at 15:39, Duraid Abbas wrote: > > > I have a D/41JCT-LS Dialogic board and I want to use it as an > > > IP-PBX. I'm new to IP Telephony and telephony and general and I > > > researched a lot but still confused about what I really need. > > > > > > I know that I can setup an IP-Telephony for my LAN using a SIP > > > server and SIP compatible software phones. But the > challenge is how > > > can I connect to the PSTN so that I can send and receive calls? > > > > Asterisk will do a wonderfull job as a soft PBX, but my > advice is to > > use hardware from Digium to connet to the PSTN (FXO or > T1/E1) and to > > connect regular analog phones (FXS or T1/E1+ChannelBank): > > > > http://www.digium.com/index.php?menu=hardware_products > > > > Before purchasing hardware, you can try to set up Asterisk > just with > > SIP softphones and get it to know the platform. Once you are > > comfortable you can jump on buying some hardware. > > > > If you do not have time to investigate yourself search for > "Asterisk > > consultants" on http://www.voip-info.org > > > > Best regards, > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/aster> isk-users > To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite to Asterisk through NAT?
On Thu, 2004-07-29 at 14:43, programmer_ted wrote: > > Wolverine looks OK, but we aren't in a position to set up another box > yet (the NAT is a router). I've set up PoPToP on the Linux box and > I'm able to connect to it from another machine fine, but we need the > same Linux box to be able to connect to it. Unfortunately, both > pptpclient and PoPToP operate on the same (non-configurable) port, so > the client can't connect to the server! > > Any ideas with my short elaboration in mind? :) OpenVPN http://openvpn.sourceforge.net/ I used it to replace traditional IPSec-based VPNs, it runs circles around them. -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RAID affecting X100P performance...
On Wed, 2004-07-21 at 12:14, Mike Benoit wrote: > I have a P3-800 with two IDE drives in a software RAID1 configuration. > Each drive is on a separate IDE channel. Now anytime there is HD > activity, I hear "beeps" and "cutting out" on a call using the X100P > card. Wow, i'm seeing exactly the same behaviour! AthlonXP/1800, MSI NForce1 mobo, Wildcard TDM400P, soft RAID1 on /boot, soft RAID5 on everything else, Asterisk-1.0-RC1, Linux Fedora 2 fully updated. I'll explore the idea offered by someone else in this thread and shuffle the cards around, trying to put the Wildcard in another PCI bus. -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
