Re: [Asterisk-Users] Callmanager 4.1 and Asterisk
i apreciatte if u can send me the conf files, and the screenshots about the CM config, its really easy as you said, i like asterisk very much, after that we are planning to make test on echo and relay calls, but think it would work great, thanx for your help, Edgar You need to create a SIP trunk in CCM and in Asterisk a peer in sip.conf with the IP address of the CCM (trunk) In the trunk configuration change the transport to UDP. Enter the IP of Asterisk. And create a route pattern with gateway the SIP trunk In Asterisk in extensions.conf create the route to CCM phones. I have this setup in my lab with CCM 4.02sr1 and works so fine. If you need the sip.conf / extensions.conf and an screenshot of the route pattern and SIP trunk config just let me know! Happy holidays! Keith O'Brien [EMAIL PROTECTED] wrote: I have a similar setup. To make it easy and get the best of both worlds, have the Linux softphones (SIP or IAX) register to Asterisk. Keep the physical phones registered to CM. From there setup a dialplan on both Call Manager and As terisk to relay calls between the two systems. For example, assign all physical phones extension 2XXX and softphones 3XXX. Have asterisk route 2XXX calls to CM via SIP and vice versa on Call Manager. Also, just so that you are aware you can register a SIP Linux softclient to Cisco Call Manager if you are running Version 4.1 --- Hello everybody, im newbie in VoIP, but find this project asterisk very interesting, i tried to install and its a great sw, i really get sorprised about all of its functions, we need to use the asterisk server in conjunction with cisco callmanager. We have a Cisco Callmanager 4.1 and the clients are softphones from cisco IPCommunicator, but all the support service of our company are linux machines, i read about callmanager uses skinny a propetary protocol and there are no softphones from linux to talk with it, so we need to install vmware to use ipcommunicator or the other solutions as i read is get the asterisk server using sip phones in the linux and windows machines and configure the call manager to talk with the asterisk server thru sip protocol, is this the real way to do that?? is there a easy way to do this?? i found this link http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration but i need to know what things to do to transfer all the extensions from de callmanager to the asterisk sw, or if only made the changes in the sip.conf as said in the link above the callmanager gets all the control?? or if i need to declare all the extensions in the asterisk?? can anybody help me?? TIA Edgar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more.___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp-0.0.2pre6
Just wondering if anyone else has managed to successfully compile spandsp-0.0.2pre6... I'm assuming it is working for most people, else it probably wouldn't have been released, and/or, other people would have had more to say about it... In any case, these are the errors I am getting while doing a make: if /bin/sh ../libtool --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I. -I-g -O2 -MT t30.lo -MD -MP -MF .deps/t30.Tpo -c -o t30.lo t30.c; \ then mv -f .deps/t30.Tpo .deps/t30.Plo; else rm -f .deps/t30.Tpo; exit 1; fi gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -MT t30.lo -MD -MP -MF .deps/t30.Tpo -c t30.c -fPIC -DPIC -o .libs/t30.o gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -MT t30.lo -MD -MP -MF .deps/t30.Tpo -c t30.c -o t30.o /dev/null 21 if /bin/sh ../libtool --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I. -I-g -O2 -MT t31.lo -MD -MP -MF .deps/t31.Tpo -c -o t31.lo t31.c; \ then mv -f .deps/t31.Tpo .deps/t31.Plo; else rm -f .deps/t31.Tpo; exit 1; fi gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -MT t31.lo -MD -MP -MF .deps/t31.Tpo -c t31.c -fPIC -DPIC -o .libs/t31.o t31.c:60: unknown field `s_regs' specified in initializer t31.c:61: unknown field `s_regs' specified in initializer t31.c:62: unknown field `s_regs' specified in initializer t31.c:63: unknown field `s_regs' specified in initializer t31.c:64: unknown field `s_regs' specified in initializer t31.c:65: unknown field `s_regs' specified in initializer t31.c:66: unknown field `s_regs' specified in initializer make[2]: *** [t31.lo] Error 1 make[2]: Leaving directory `/usr/src/asterisk/spandsp-0.0.2/src' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/asterisk/spandsp-0.0.2/src' make: *** [all-recursive] Error 1 I've tried various 'hacks' to the source code to try and make it compile, but I don't know enough C to successfully do even that... Anyone got any suggestions? Is there something else I should look for?? Thanks, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Invalid Extension
I can dial-in and here the prompt, but whenever I select 101, I get invalid extension. May I ask, is this the right way of answering incoming calls? I had to change all occurance of s to 533990 in order for this to work. 533990 is my FWD #. May I ask how can I genearlize this using s? Regards, Norman Zhang [inbound-sip] exten = 533990,1,Answer exten = s,2,ResponseTimeout(5) exten = s,3,Background(mymenu) exten = t,1,Goto(s,2) exten = i,1,Playback(pbx-invalid) exten = i,2,Goto(s,2) exten = 101,1,Goto(local,101,1) exten = 138,1,Goto(local,138,1) When you register with FWD, you used something like: register=userid:[EMAIL PROTECTED]/533990 where you've included /533990 at the end. That is telling FWD what exten number to send to your * box when receiving a call. Remove that and the 's' extension will work just fine. The 's' extension is a special start case that does not expect any digits to be passed to it from FWD in this case. So, you can use either approach in the dialplan, but you need to be consistent throughout. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to debug frame slips?
On Wed, 29 Dec 2004, Adam Goryachev wrote: On Tue, 2004-12-28 at 06:10, Michael Welter wrote: Try 'lspci -v' and look at the latency timer for your Digium card(s). You can set it higher with 'setpci -v -s xx:yy.0 LATENCY=TIMER=ff' (xx is the bus number and yy is the slot). Shouldn't you decrease the latency? ie, to something lower like 16 or 8 or 0 ?? or is a higher value better?? Generally, you want the digium cards to have a high enough latency value on the digium cards that it can transfer one ms worth of audio in one burst (8*numchans bytes). You also want other cards to have low enough latency timers to grant the digium cards access to the bus soon enough, typically a fraction of a ms. See http://www.reric.net/linux/pci_latency.html. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Final call for departments
I am getting ready to submit a list of department names to be recorded. This is what I have so far: Accounting Accounts payable Accounts receivable Administration Billing Collections Complaint Customer Service Engineering Facilities Help desk Human Resources Information Technology Inside Sales Investor Relations Legal Mail room Marketing Printing Projects Public Relations Purchasing Receiving Sales Sales Floor Shipping Shop Support Systems Technical Support Travel If any one has additional suggestions, please e-mail them to me ([EMAIL PROTECTED] or [EMAIL PROTECTED]). I am fairly sure that none of the above exist (I was only able to search through the WIKI list, so if there are other prompts in the CVS that are not listed there, I do not know about them.) If I have made a dupe, please let me know so that I can remove it. I was fairly certain that 'Operator' was already available but I was unable to find it by its self. Thanks for your help. I plan on sending these off on Friday the 31st so please try to get them to me by then. Thanks; James ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Determine UAS on remote SIP phones
Hi all This question is not connected directly wiht Asterisk but I didn't find another place. Sorry, I would like to determine state of UA ( DND, forward calls turned on my remote sip phones.) Which tools should I use ? sipsak ? maybe something else. When I send options method to remote sip phones I always receive SIP/2.0 200 OK response regardless of activated call features (DND, forward) How can I distinguish different state of UAS ? Thansk radan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip phone configuration problem
733 is a remote IP phone 608 is X-Lite on the internal LAN I see a: *CLI... Registration from '733 sip:[EMAIL PROTECTED]' failed for '218.x.x.x' *CLI sip show peers 733/733(Unspecified ) D 255.255.255.2550Unknown 608/608192.168.250.200D 255.255.255.255 5060OK (26 ms) *CLI sip show users 733 password test733 No No 608 password default No No I can call 608, but 608 cannot call back 733 cannot register below is my sip.conf, ... Can anybody give me a hint? sip.conf [general] context=defaultport=5060 bindaddr=0.0.0.0 srvlookup=yes externip = 61.220.121.18 localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks [608] ; Note-Pen's X-lite type=friend disallow=all allow=ulaw allow=alaw type=friend username=608 secret=password host=dynamic dtmfmode=inband qualify=1000 mailbox=608 group=1 pickupgroup=1 [733] ; Test phone 733 context=unisen disallow=all allow=ulaw allow=alaw type=friend username=test733 secret=password host=dynamic dtmfmode=inband qualify=1000 mailbox=733 group=1 pickupgroup=1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] API Manager Events
Hi, I'm trying to develop a java web application that places calls using Asterisk. Basically it has to ring two terminals and connect them. Our first approach was generating a .call file and place it in the spool directory. Later we found out that our app needs to know what calls are currently held and their duration, in other words it has to know when the placed calls are terminated. We then thought about using the Manager API, and control the event flow to track the status of the calls. Here are my questions: 1) is there eny documentation on the events asterisk may fire? 2) is there any way to send an especific custom event to the Manager API from the dialplan(and forget about the rest of events)? any other idea will be appreciated. Thanks in advance pressec ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with musiconhold - No such file or directory
I have mpg123 v59r and in command line mpg123 play of music /var/lib/asterisk/mohmp3/sample.mp3 in musiconhold.conf I have: -- [classes] default = mp3:/var/lib/asterisk/mohmp3/ - and extensions.conf exten = 2,1,WaitMusicOnHold(30) When I call on console asterisk I have message: - -- Executing WaitMusicOnHold(SIP/100-2278, 30) in new stack -- Started music on hold, class 'default', on SIP/100-2278 sample.mp3: No such file or directory - Why asterisk don't see my sample.mp3 ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE:[Asterisk-Users] Problems with loading TE110 module
Title: Mensaje Have you solve your Problem?, I have same problem after with recompile kernel. Regards, srsergio Monday, December 20, 2004, 12:44:36 PM, Matt wrote:MR Have you tried doing a modprobe -r first?Before reboot I did rmmod wcte11xp. If you mean that.now modprobe -r wcte11xp doesn't do anything, still can't load themodule. :(Tamas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE110P doesn't appear in /proc/zaptel
Title: Mensaje Hi all, I have installed a TE110P in a BOX but when I load zaptel module I can't see any device in /proc/zaptel. And led of the card is green. My zaptel.conf is the next: span=1,1,0,ccs,hdb3,crc4bchan=1-15,17-31dchan=16loadzone=esdefaultzone=es and cat /proc/pci throguh next: PCI devices found: Bus 0, device 0, function 0: Host bridge: Intel Corp. 82845 845 (Brookdale) Chipset Host Bridge (rev 4). Prefetchable 32 bit memory at 0xd000 [0xd7ff]. Bus 0, device 1, function 0: PCI bridge: Intel Corp. 82845 845 (Brookdale) Chipset AGP Bridge (rev 4). Master Capable. Latency=64. Min Gnt=14. Bus 0, device 30, function 0: PCI bridge: Intel Corp. 82801BA/CA/DB PCI Bridge (rev 5). Master Capable. No bursts. Min Gnt=6. Bus 0, device 31, function 0: ISA bridge: Intel Corp. 82801BA ISA Bridge (LPC) (rev 5). Bus 0, device 31, function 1: IDE interface: Intel Corp. 82801BA IDE U100 (rev 5). I/O at 0xf000 [0xf00f]. Bus 0, device 31, function 2: USB Controller: Intel Corp. 82801BA/BAM USB (Hub #1) (rev 5). IRQ 10. I/O at 0xd000 [0xd01f]. Bus 0, device 31, function 3: SMBus: Intel Corp. 82801BA/BAM SMBus (rev 5). IRQ 9. I/O at 0x500 [0x50f]. Bus 0, device 31, function 4: USB Controller: Intel Corp. 82801BA/BAM USB (Hub #2) (rev 5). IRQ 12. I/O at 0xd800 [0xd81f]. Bus 0, device 31, function 5: Multimedia audio controller: Intel Corp. 82801BA/BAM AC'97 Audio (rev 5). IRQ 9. I/O at 0xdc00 [0xdcff]. I/O at 0xe000 [0xe03f]. Bus 1, device 0, function 0: VGA compatible controller: ATI Technologies Inc Radeon VE QY (rev 0). IRQ 5. Master Capable. Latency=32. Min Gnt=8. Prefetchable 32 bit memory at 0xd800 [0xdfff]. I/O at 0xc000 [0xc0ff]. Non-prefetchable 32 bit memory at 0xe100 [0xe100]. Bus 2, device 1, function 0: Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8029(AS) (rev 0). IRQ 10. I/O at 0xa000 [0xa01f]. Bus 2, device 5, function 0: Network controller: Tiger Jet Network Inc. Model 300 128k (rev 0). IRQ 12. Master Capable. Latency=32. Min Gnt=1.Max Lat=128. I/O at 0xa400 [0xa4ff]. Non-prefetchable 32 bit memory at 0xe300 [0xe3000fff]. Any idea? regards, srsergio -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] caller-id blocking
Hi; How can a user block his caller-id in Astersik? Regards Mohammad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
I'm guessing he wants to do it the other way around, i.e. the external calling party hears music, not the internal calling party making an external call. -Original Message- From: Peter Svensson [mailto:[EMAIL PROTECTED] Sent: 28 December 2004 21:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Music instead of Tunes On Tue, 28 Dec 2004, Marc Storck wrote: more and more operators in Europe offer music instead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart Currently I will have to answer the line to do that. Is there a way to do this with asterisk? See the help for dial: 'm' -- provide hold music to the calling party until answered. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk handle calls that get picked upby answering machines?
How do those telemarketers do it? I come home and I got messages on my answerings machine that are recordingsof course thats not what I want to do...but its handing answerings machines is something possible. Does Asterisk not have any feature or tool or anything for this? - Original Message - From: Todd Lieberman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 25, 2004 7:06 AM Subject: Re: [Asterisk-Users] Can Asterisk handle calls that get picked upby answering machines? Gabriel Afana wrote: Just wondering because right now I can have it call my phone and play a message, but if I dont answer it eventually goes to voice mail. It always leaves a voicemail and when I listen to it its always the last few seconds of my message that I had Asterisk play. How do I get Asterisk to pause and wait to playback *IF* its an answering machine or voicemail? Gabe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You can hack the record application to wait for silence. -- Todd Lieberman mailto:[EMAIL PROTECTED] http://tlsolutions.net 215.495.0030 (p) 215.495.0031 (f) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] API Manager Events
On Wed, 29 Dec 2004, Dani wrote: 2) is there any way to send an especific custom event to the Manager API from the dialplan(and forget about the rest of events)? See the dialplan application UserEvent. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
On Wed, 29 Dec 2004, Steve Hanselman wrote: On Tue, 28 Dec 2004, Marc Storck wrote: more and more operators in Europe offer music instead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart Currently I will have to answer the line to do that. Is there a way to do this with asterisk? See the help for dial: 'm' -- provide hold music to the calling party until answered. I'm guessing he wants to do it the other way around, i.e. the external calling party hears music, not the internal calling party making an external call. Are the two cases different in any way? The external call comes in, goes to a context which eventually leads to a Dial(...) calling the internal user. That Dial call provides music to the external caller while the internal call is in progress. Asterisk has no concept of external or internal callers, only channles and contexts. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZtDummy vs Hardware
So, 215 - 40 - 110 = 65kbps You only have about 65kbps to spare, and all of this is based on ideal (theoritical) conditions. I doubt that those 12 calls will sound okay, or even work at all... But, you can always try! The thing is: how do I do that? What tools are there to test how many channels can be safely fit through? Another thing is that I was thinking that it would be possible to save bandwith by lengthening the frame length to 50 or even 90ms. I have no idea how this works or how to do it though :( Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Impossible to compile last version of Asterisk
Hi, I worked with Asterisk 0.7 without problems until I tryed to loadH323. I downloaded the last version and after some try I compile it.I followed the description in /asterisk/channels/h323/Readmeand the compilation of this part was good. But the new compilationof Asterisk was impossible (problem with chan_h323.so). I searchinfo with Google and I read that the problem could be with the different kind of version.Today I downloaded the last version of Zaptel, Libpri and Asterisk fromcvs and I followed the description in http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation.For Zaptel and Libpri it's ok. but I can' t compile Asterisk...somebody can help me? Thank you very muchmy version of kernel, I've SUSE 8.1linux:/usr/src/asterisk # uname -aLinux linux 2.4.21-238-default #1 Thu Jul 29 17:37:30 UTC 2004 i686 unknownlinux:/usr/src/asterisk #linux:/usr/src/asterisk # cat /proc/versionLinux version 2.4.21-238-default ([EMAIL PROTECTED]) (gcc version 3.2.2) #1 Thu Jul 29 17:37:30 UTC 2004linux:/usr/src/asterisk #the errormake[1]: Leaving directory `/usr/src/asterisk/cdr'make[1]: Entering directory `/usr/src/asterisk/utils'gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-HEAD-12/29/04-09:53:32\" -DASTERISK_VERSION_NUM=99 -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -DNO_AST_MM -c -o smsq.o smsq.csmsq.c: In function `main':smsq.c:422: `POPT_ARGFLAG_SHOW_DEFAULT' undeclared (first use in this function)smsq.c:422: (Each undeclared identifier is reported only oncesmsq.c:422: for each function it appears in.)make[1]: *** [smsq.o] Error 1make[1]: Leaving directory `/usr/src/asterisk/utils'make: *** [subdirs] Error 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending e-mail from dialplan
AGI Script. I would like help with a dial plan that will do the following: I feel pretty good about each of the elements except; how to e-mail the recorded file to an e-mail address. Allow a caller to call into the system: 1.Answer 2.play a short pre defined greeting 3.Allow caller to enter PIN during the Item #2 greeting a. If the caller entered THE valid pin (1 system wide pre-defined pin) the caller she experience: i. Be prompted to record a greeting (record action mandatory) defined as the message of the day ii. Listen to recorded greeting for approval, option to re-record or option to continue. iii. After continuing, the caller should have an option to send message to the E-mail list as a .WAV attachment. 1. The Email list will be a single address to a mail server for distribution to member lists. iv. Thank the caller v. Disconnect b. If the caller does not enter a PIN in 15sec the caller is played the Current time and Date, a recorded disclaimer, and the Message of the day, then is disconnected. Also, while you're on the topic; what is the feasibility of allowing someone to hit an web page and type in the message of the day and have festival read it? If that may work, I would send the e-mail from the web page action and asterisk would not have to handle it But for simplicity and end user ease over the phone would be better I think? Thanks for all of your help, AZM The Labs -- Brian Wilkins [EMAIL PROTECTED] Software Engineer Heritage Communications Corporation Melbourne, FL USA 32935 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk OH323 acting as a gatekeeper
Hi, I am testing asterisk in several situations, I still can not understand why we have to deal with two different h323 channels... Here is the problem, I have a cisco 3700 who sends h323 calls to asterisk. then I process the call upon several users parameters, and send it to another cisco gateway. All the transactions are made in h323. I first tried with h323 channel, and had no audio... I guess that is because g729 is not g729a. So I tried with oh323, and I finally got audio! Unfortunately the oh323 acts like a gatekeeper... and trys to create the direct call between the two ciscos... therefore, once the communication is accomplish it hangs up. I had to make available both g729a anf g729 in my oh323.conf, since the first cisco appears to use that codec. I tried all the combinations successfully, since cisco-asterisk-sip works ok and asterisk-cisco2 works too. I am sure this works since asterisk is transcoding can not make the rtp to flow directly between the two clients. Does somebody knows how to deal with this? Thanks, Alito ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZtDummy vs Hardware
So, 215 - 40 - 110 = 65kbps You only have about 65kbps to spare, and all of this is based on ideal (theoritical) conditions. I doubt that those 12 calls will sound okay, or even work at all... But, you can always try! The thing is: how do I do that? What tools are there to test how many channels can be safely fit through? The only way to know for sure is to try it in a test environment. Place 6, 10, 12 calls to someone else's asterisk box, measure the bandwidth consumed, and listen to the quality for one of the calls in each test. Lots of free tools out there to help measure/display the actual bandwidth consumed. And, you certainly don't need much of a system to test this, but you would need to commit to the dsl circuit obviously. Doing the pencil calculations is fine for starters, but there are lots of other parameters that can impact the pencil-best-case such as propagation delays, ISP bandwidth throttling (for many different reasons), incorrect half vs full duplex settings anywhere along the entire path, dsl modem irregularities, etc. You might also find that iax trunking doesn't work the way you thought it would on paper, etc. Another thing is that I was thinking that it would be possible to save bandwith by lengthening the frame length to 50 or even 90ms. I have no idea how this works or how to do it though :( That might be a consideration _after_ you've confirmed the tests noted above, but you are really talking about some rather small incremental bandwidth improvements. (Particularly if you use iax trunking, which I think you mentioned in an earlier post.) Since changing the frame length is not something that a lot of people actually try, you're also likely to stumble across coding errors, etc, that have not yet been uncovered. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
The difference is that you'd have to answer the call, my guess is that it can't be done (by a Joe Average like ourselves), otherwise we'd provide useful information to callers at no charge. -Original Message- From: Peter Svensson [mailto:[EMAIL PROTECTED] Sent: 29 December 2004 11:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Music instead of Tunes On Wed, 29 Dec 2004, Steve Hanselman wrote: On Tue, 28 Dec 2004, Marc Storck wrote: more and more operators in Europe offer music instead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart Currently I will have to answer the line to do that. Is there a way to do this with asterisk? See the help for dial: 'm' -- provide hold music to the calling party until answered. I'm guessing he wants to do it the other way around, i.e. the external calling party hears music, not the internal calling party making an external call. Are the two cases different in any way? The external call comes in, goes to a context which eventually leads to a Dial(...) calling the internal user. That Dial call provides music to the external caller while the internal call is in progress. Asterisk has no concept of external or internal callers, only channles and contexts. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] caller-id blocking
On December 28, 2004 06:32 pm, mohammad wrote: How can a user block his caller-id in Astersik? show application SetCallerPres -= Info about application 'SetCallerPres' =- [Synopsis]: Set CallerID Presentation [Description]: SetCallerPres(presentation): Set Caller*ID presentation on a call to a new value. Sets ANI as well if a flag is used. Always returns 0. Valid presentations are: allowed_not_screened: Presentation Allowed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen: Presentation Prohibited, Passed Screen prohib_failed_screen: Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable You could also use SetCIDName/SetCIDNum as a more brute-force method. Note these likely only work on ISDN BRI/PRI interfaces. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel ISDN BRI settings for The Netherlands KPN
Hello (Hallo) Remco Barende Just look at http://www.ip-phone-forum.de/forum/portal.php it's in German language but for a dutch guy no problem I gues ? Sjaak Hi list! I am installing an * box that will be installed on a site with KPN BRI ISDN in The Netherlands. I am using bristuff fron Junghanns. Does anybody know the correct settings for this? I will not have internet access there which makes it harder to google around on location. switchtype = euroisdn is pretty obvious but what about these settings: signalling = bri_cpe_ptmp ; p2p TE mode ;signalling = bri_cpe ; p2mp NT mode ;signalling = bri_net_ptmp ; p2p NT mode ;signalling = bri_net pridialplan=local prilocaldialplan=local ; trust user provided callerid (clip no screening)? pritrustusercid = yes immediate=yes channel = 1-2 Thanks!! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AstTAPI - Incoming Calls
Good day, does anyone have AstTAPI running for incoming calls, and would like to show some examples. My setting right now looks like this: sip.conf [22] type=friend dtmfmode=info username=22 mailbox=22 secret=privat host=dynamic context=privat canreinvite=yes callgroup=1 incominglimit=2 extension.conf -- exten = 123,1,noop ;Hint(SIP/22) exten = 123,2,Dial(SIP/22,20,t) exten = 123,3,Voicemail2(su22) exten = 123,4,Hangup exten = 123,103,VoiceMail2(su22) exten = 123,104,Hangup The TAPI settings look like this: User channel: Sip/22 Inbound Chan: Sip/22 The manager logs in just fine using the Windows 2000 Dialer App and does logout until I exit the application. I can dial out, using the context defined in Dial by Context. However, after specifying the inbound channel, no more calls get thru to the Sip/22 extension. Here is an extract of the log: == Parsing '/etc/asterisk/manager.conf': Found == Manager 'pos' logged on from 10.1.3.68 == Manager 'pos' logged off from 10.1.3.68 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'pos' logged on from 10.1.3.68 -- Executing NoOp(CAPI[contr3/123]/204, ) in new stack -- Executing Dial(CAPI[contr3/123]/204, SIP/22|20|t) in new stack -- Called 22 -- SIP/22-4eec is ringing == No one is available to answer at this time -- Executing VoiceMail2(CAPI[contr3/123]/204, su22) in new stack -- Playing 'voicemail/default/22/unavail' (language 'en') == Spawn extension (default, 123, 3) exited non-zero on 'CAPI[contr3/123]/204' The phone rings once. Nobody used the phone at that time, but the call gets directed into voice mail and no pop up happens on the Windows client. Using Outlook 2000 the manager does not log in unless I want to Dial-Out. And it logs out right after completing the call successfully. Any and all help is greatly appreciated. -- Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL Realtime Driver
Maybe you could point me to the _detailed_ instructions on the wiki. I appreciate the work that's been done. I'd just like to continue it via documentation. A SQL dump or at least a table schema would be beneficial as would be a sample configuration. On Tue, 28 Dec 2004 22:29:29 -0600, Matthew Boehm [EMAIL PROTECTED] wrote: There is info on the wiki. Matthew - Original Message - From: Chris Tooley [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, December 28, 2004 5:38 PM Subject: Re: [Asterisk-Users] MySQL Realtime Driver Is there any documentation or insight on figuring out how to get RealTime IAX set up? I'm trying to do just that. Also can do separate peer/user configurations or just friends? And how do you configure the rest of the iax.conf information? On Fri, 10 Dec 2004 10:01:00 -0600, Matthew Boehm [EMAIL PROTECTED] wrote: Yes and No. You need to realize this isn't Asterisk and MySQL. This is Asterisk and RealTime using MySQL. You can also have Asterisk and RealTime using ODBC etc.. It is NOT the database that supports features. It is RealTime that supports features. RealTime is still in DEVELOPMENT. and more apps are slowly being added with RealTime abilities. Currently, the only officially supported RealTime configs are sipfriends, iaxfriends, voicemail and extensions. There are patches in progress for MeetMe, and Directory. Yes, you can store static *.confs into the database just like before. You need to be running latest CVS. If you want to use ODBC-MySQL then you don't need anything extra. If you want direct MySQL, get (from CVS) asterisk-addons. -Matthew - Original Message - From: Christopher Jacob [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, December 10, 2004 8:22 AM Subject: [Asterisk-Users] MySQL Realtime Driver Can someone shed some light on this? It sounds like exactly what I am looking for. Does it handle extensions.conf or just sip/iax/voicemail? (not that to say that _just_ those things would be cool) I have googled for some more information, but so far the only thing I can find is in the bug tracker and perhaps I'm missing something, but I don't get a full explanation. Any insight would be greatly appreciated. ~c ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Linux routing with T100P problems
Check with your ISP and make sure they have you set up correctly. I have had issues in the past with that. What's funny is that I am the ISP. There is nothing between the two T1 endpoints. BTW, I can ping the above IP from my machine just fine, so the rest of the world sees your T1 as well. Haha. Well the IP addresses I put in the diagram are not mine, I changed them to protect the innocent but.. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Linux routing with T100P problems
On Dec 29, 2004, at 12:18 AM, Matthew Boehm wrote: Hey gang, I was successful in recompiling my 2.4.20 kernel to support HDLC. I was successful in hooking up our T1 line into the zap card. I was successful in being able to ping equipment on the other end of the T1. I was unsuccessful in pinging the outside world from the other end of the T1. I've attached a cheezy image of the network. Here is the routing table: [EMAIL PROTECTED] root]# route Kernel IP routing table Destination Gateway Genmask Flags Metric RefUse Iface 10.0.5.2 * 255.255.255.255 UH0 0 0 hdlc0 10.0.0.0 * 255.255.255.0 U 0 0 0eth1 10.0.3.0 * 255.255.255.0 U 0 0 0eth1 65.78.109.0 * 255.255.255.0 U 0 0 0 eth0 127.0.0.0 * 255.0.0.0 U 0 0 0lo default 65.78.109.2 0.0.0.0 UG0 0 0eth0 There are 2 NICs (10.0.3.10, 65.78.109.10) and 1 T100P (10.0.5.1) on this box. Like I said above, from this machine I can ping everything in every attached network and the outside world. For some reason, I cannot ping the outside world if I am comming from the 10.0.0.* network on the diagram. From that network, I can ping 10.0.5.1 (this box) but nothing else. appears that your box isn't configured for NAT, so you want to brush up on iptables. Most distributions make this pretty easy, and of course each distro has a different approach on where to find the preconfigured scripts. (google) Niles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Linux routing with T100P problems
And your cheezy network image shows you have not exhibited good networking knowledge. You show an internet clod in the middle of a point to point T1. That cloud pic only serves to show that there is a 30 mile gap between the two endpoints. Why is it you have 10.0.0.2 as a IP on the other end of a router on the T1 line and you are routing it out of the eth1 device. router1 is the main office router. All equipment in the office is in the 10.0.0.* network. And so the eth interface of router1 is 10.0.0.2. eth1 on the * box actually goes to a switch that connects to a PIX. I can send 10.0.*.* traffic out eth1 and the PIX routes it. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL Realtime Driver
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime http://www.voip-info.org/wiki-Asterisk+RealTime+IAX Since I don't use IAX peers/users and since nobody has bothered to update the IAX page with a database schema, you are on your own on that aspect. -matthew - Original Message - From: Chris Tooley [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 29, 2004 8:02 AM Subject: Re: [Asterisk-Users] MySQL Realtime Driver Maybe you could point me to the _detailed_ instructions on the wiki. I appreciate the work that's been done. I'd just like to continue it via documentation. A SQL dump or at least a table schema would be beneficial as would be a sample configuration. On Tue, 28 Dec 2004 22:29:29 -0600, Matthew Boehm [EMAIL PROTECTED] wrote: There is info on the wiki. Matthew - Original Message - From: Chris Tooley [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, December 28, 2004 5:38 PM Subject: Re: [Asterisk-Users] MySQL Realtime Driver Is there any documentation or insight on figuring out how to get RealTime IAX set up? I'm trying to do just that. Also can do separate peer/user configurations or just friends? And how do you configure the rest of the iax.conf information? On Fri, 10 Dec 2004 10:01:00 -0600, Matthew Boehm [EMAIL PROTECTED] wrote: Yes and No. You need to realize this isn't Asterisk and MySQL. This is Asterisk and RealTime using MySQL. You can also have Asterisk and RealTime using ODBC etc.. It is NOT the database that supports features. It is RealTime that supports features. RealTime is still in DEVELOPMENT. and more apps are slowly being added with RealTime abilities. Currently, the only officially supported RealTime configs are sipfriends, iaxfriends, voicemail and extensions. There are patches in progress for MeetMe, and Directory. Yes, you can store static *.confs into the database just like before. You need to be running latest CVS. If you want to use ODBC-MySQL then you don't need anything extra. If you want direct MySQL, get (from CVS) asterisk-addons. -Matthew - Original Message - From: Christopher Jacob [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, December 10, 2004 8:22 AM Subject: [Asterisk-Users] MySQL Realtime Driver Can someone shed some light on this? It sounds like exactly what I am looking for. Does it handle extensions.conf or just sip/iax/voicemail? (not that to say that _just_ those things would be cool) I have googled for some more information, but so far the only thing I can find is in the bug tracker and perhaps I'm missing something, but I don't get a full explanation. Any insight would be greatly appreciated. ~c ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Linux routing with T100P problems
appears that your box isn't configured for NAT, so you want to brush up on iptables. Ahhh. Didn't think about that. Makes sense now..if I send a packet to the outside world how is it going to get back to the originator when the originator is inside a 10.0.*.* network. I forgot that we have a hardware NAT device that normally handles that. Thanks, Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Linux routing with T100P problems
On Tue, Dec 28, 2004 at 11:18:42PM -0600, Matthew Boehm wrote: Hey gang, I was successful in recompiling my 2.4.20 kernel to support HDLC. I was successful in hooking up our T1 line into the zap card. I was successful in being able to ping equipment on the other end of the T1. I was unsuccessful in pinging the outside world from the other end of the T1. I've attached a cheezy image of the network. Here is the routing table: [EMAIL PROTECTED] root]# route Kernel IP routing table Destination Gateway Genmask Flags Metric RefUse Iface 10.0.5.2 * 255.255.255.255 UH0 00 hdlc0 10.0.0.0 * 255.255.255.0 U 0 0 0eth1 10.0.3.0 * 255.255.255.0 U 0 0 0eth1 65.78.109.0 * 255.255.255.0 U 0 00 eth0 127.0.0.0 * 255.0.0.0 U 0 0 0lo default 65.78.109.2 0.0.0.0 UG0 0 0eth0 There are 2 NICs (10.0.3.10, 65.78.109.10) and 1 T100P (10.0.5.1) on this box. Like I said above, from this machine I can ping everything in every attached network and the outside world. For some reason, I cannot ping the outside world if I am comming from the 10.0.0.* network on the diagram. From that network, I can ping 10.0.5.1 (this box) but nothing else. I'm a little stumped. My iptables are completly empty. If this is waaayyy off topic, please contact me off list. But I figured since it was related to the T100P it might be relevant. What can I use to find out why packets destined for the outside world (via 65.78.109.2) are not being routed? Since 10.x.x.x is RFC1918 private space which no real-world addresses will/can reply to, you need to use masquerading (NAT) so that all of the packets to the outside world appear to come from a public routable address on the outside of your gateway box. -Dorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycomm IP500 dropping incoming calls
Hello everyone. I can place outgoing calls no problem with my IP500 (using teliax as our provider). Thing is, when a call comes in, 90% of the time when I pick up the handset it drops the call immediately. I turned on SIP debug, and have listed my extension config from sip.conf. Any help is greatly appreciated sooo close TIA! -Ron [3004] type=friend username=3004 password=XXX host=dynamic ;host=192.168.4.204 ;host=static dtmfmode=inband defaultip=192.168.4.204 context=default disallow=all allow=ulaw ;nat=yes callerid=George W. Bush 3004 mailbox=3004 SIP Debugging Enabled -- Accepting AUTHENTICATED call from 204.188.109.139, requested format = 4, actual format = 4 -- Executing DigitTimeout([EMAIL PROTECTED]/3, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout([EMAIL PROTECTED]/3, 10) in new stack -- Set Response Timeout to 10 -- Executing Macro([EMAIL PROTECTED]/3, stdexten|3004|SIP/3004) in new stack -- Executing Dial([EMAIL PROTECTED]/3, SIP/3004|20) in new stack We're at 192.168.4.5 port 15760 Answering with preferred capability 4 12 headers, 8 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf From: 3124048745 sip:[EMAIL PROTECTED];tag=as5e966399 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 29 Dec 2004 20:20:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 156 v=0 o=root 1879 1879 IN IP4 192.168.4.5 s=session c=IN IP4 192.168.4.5 t=0 0 m=audio 15760 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - (no NAT) to 192.168.4.204:5060 -- Called 3004 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf From: 3124048745 sip:[EMAIL PROTECTED];tag=as5e966399 To: sip:[EMAIL PROTECTED];tag=EAA91427-3070A3C8 CSeq: 102 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 9 headers, 0 lines Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf From: 3124048745 sip:[EMAIL PROTECTED];tag=as5e966399 To: sip:[EMAIL PROTECTED];tag=EAA91427-3070A3C8 CSeq: 102 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Allow-Events: talk,hold,conference Content-Length: 0 10 headers, 0 lines -- SIP/3004-5a28 is ringing Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf From: 3124048745 sip:[EMAIL PROTECTED];tag=as5e966399 To: sip:[EMAIL PROTECTED];tag=EAA91427-3070A3C8 CSeq: 102 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Type: application/sdp Content-Length: 148 v=0 o=- 915180542 915180542 IN IP4 192.168.4.204 s=Polycom IP Phone c=IN IP4 192.168.4.204 t=0 0 m=audio 2236 RTP/AVP 0 a=rtpmap:0 PCMU/8000 11 headers, 7 lines Found audio format UNKN Found description format PCMU Capabilities: us - 4, them - 4/0, combined - 4 Non-codec capabilities: us - 1, them - 0, combined - 0 list_route: hop: sip:[EMAIL PROTECTED]:5060 set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 192.168.4.204, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf From: 3124048745 sip:[EMAIL PROTECTED];tag=as5e966399 To: sip:[EMAIL PROTECTED];tag=EAA91427-3070A3C8 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.4.204:5060 -- SIP/3004-5a28 answered [EMAIL PROTECTED]/3 set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 192.168.4.204, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf From: 3124048745 sip:[EMAIL PROTECTED];tag=as5e966399 To: sip:[EMAIL PROTECTED];tag=EAA91427-3070A3C8 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.4.204:5060 == Spawn extension (macro-stdexten, s, 1) exited non-zero on '[EMAIL PROTECTED]/3' in macro 'stdexten' == Spawn extension (default, 9722150488, 3) exited non-zero on '[EMAIL PROTECTED]/3' -- Hungup '[EMAIL PROTECTED]/3' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf From: 3124048745 sip:[EMAIL PROTECTED];tag=as5e966399 To: sip:[EMAIL PROTECTED];tag=EAA91427-3070A3C8 CSeq: 103 BYE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length:
RE: [Asterisk-Users] Music instead of Tunes
On Wed, 29 Dec 2004, Steve Hanselman wrote: Are the two cases different in any way? The external call comes in, goes to a context which eventually leads to a Dial(...) calling the internal user. That Dial call provides music to the external caller while the internal call is in progress. The difference is that you'd have to answer the call, my guess is that it can't be done (by a Joe Average like ourselves), otherwise we'd provide useful information to callers at no charge. For pots lines this is true. For isdn lines there is no need to answer prior to sending data. The reverse path (from the called party towards the calling party) is opened when (this is form memory, it may be another IE) PROGRESS is transmitted. You can use Playback and a host of other connads on an unanswered line. Some of these will automatically answer the line unless given an option not to. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware opinions?
On Tue, Dec 28, 2004 at 11:37:42PM -0600, Me wrote: What sort of chipset is your SATA controller interface? Intel ICH6R? Adaptec ICH5R SATA controller according to SuperMicro which makes the Mobo. The board has an Intel® E7501 main chipset. That should probably work. You may need to reconfigure your kernel, or maybe not. I can't say for Fedora. run make menuconfig, go into Device Drivers, then into SCSI Device Support (yes, that's where the good SATA stuff hides), then into SCSI low-level drivers (at the bottom), where you will find a section that starts with Serial ATA (SATA) Support. I am using AHCI SATA Support, which is very nice, but depends on your motherboard bios having an AHCI mode for the SATA disks. [if you can use this mode, I highly recommend it, as the performance is shockingly good] If you don't have AHCI, the Intel PIIX/ICH support may work for you. There are also drivers from various other flavors of motherboard controllers, but I haven't fooled with them. If the support you need is already built in your kernel, then you may not need to rebuild it. I recommend building the relevant drivers hard into the kernel (not loading them as modules) since you're going to need them all the time anyway. [I may be clueless on that, but it works for me :) ] Happy Holidays! -Dorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] trimming messages on reply
All, Please consider trimming off the bottom of the message you are replying to. It usually take only a few seconds and saves everyone reading the list from extra bloat in their mailbox :) Happy Holidays! -Dorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] trimming messages on reply
Will do. __ Dana -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dorn Hetzel Sent: Wednesday, December 29, 2004 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] trimming messages on reply All, Please consider trimming off the bottom of the message you are replying to. It usually take only a few seconds and saves everyone reading the list from extra bloat in their mailbox :) Happy Holidays! -Dorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Disclaimer: The information transmitted in this message is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination, or other use of or taking of any action in reliance upon this information by persons or entities other than the intended recipient is prohibited. If you received this message in error, please contact the sender and delete the material from any system. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp-0.0.2pre6
Hi Adam, You must be using a prehistoric GCC. Before 3.0, GCC didn't understand this C99 construct. Regards, Steve Adam Goryachev wrote: Just wondering if anyone else has managed to successfully compile spandsp-0.0.2pre6... I'm assuming it is working for most people, else it probably wouldn't have been released, and/or, other people would have had more to say about it... In any case, these are the errors I am getting while doing a make: if /bin/sh ../libtool --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I. -I-g -O2 -MT t30.lo -MD -MP -MF .deps/t30.Tpo -c -o t30.lo t30.c; \ then mv -f .deps/t30.Tpo .deps/t30.Plo; else rm -f .deps/t30.Tpo; exit 1; fi gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -MT t30.lo -MD -MP -MF .deps/t30.Tpo -c t30.c -fPIC -DPIC -o .libs/t30.o gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -MT t30.lo -MD -MP -MF .deps/t30.Tpo -c t30.c -o t30.o /dev/null 21 if /bin/sh ../libtool --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I. -I-g -O2 -MT t31.lo -MD -MP -MF .deps/t31.Tpo -c -o t31.lo t31.c; \ then mv -f .deps/t31.Tpo .deps/t31.Plo; else rm -f .deps/t31.Tpo; exit 1; fi gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -MT t31.lo -MD -MP -MF .deps/t31.Tpo -c t31.c -fPIC -DPIC -o .libs/t31.o t31.c:60: unknown field `s_regs' specified in initializer t31.c:61: unknown field `s_regs' specified in initializer t31.c:62: unknown field `s_regs' specified in initializer t31.c:63: unknown field `s_regs' specified in initializer t31.c:64: unknown field `s_regs' specified in initializer t31.c:65: unknown field `s_regs' specified in initializer t31.c:66: unknown field `s_regs' specified in initializer make[2]: *** [t31.lo] Error 1 make[2]: Leaving directory `/usr/src/asterisk/spandsp-0.0.2/src' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/asterisk/spandsp-0.0.2/src' make: *** [all-recursive] Error 1 I've tried various 'hacks' to the source code to try and make it compile, but I don't know enough C to successfully do even that... Anyone got any suggestions? Is there something else I should look for?? Thanks, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - Originating Network identity
In General,
The way carriers derive this information is from CIC codes (Carrier
Identification Codes) embedded in the SS7 overhead. This is used for
intercarrier compensation; which is the typical reason you'd care what
carrier originated the call.
For more info on CICs:
http://www.nanpa.com/number_resource_info/carrier_id_codes.html
It's a tricky problem to solve really in today new fangled telephony.
What do you call the originating carrier The local provider? The end
user? The wholesaler? The IXC? A call that hits you could have
traversed a dozen networks and the originating network could have been a
system as simple (or complex) as an asterisk box with one analog
interface and no SS7 capabilities. Therefore unable to populate a CIC
code. Therefore their upstream provider may populate that field.
Now, going by strict guidelines CallerID information (including callerid
number) has NEVER been and should NEVER be used for routing of calls or
rating of calls as this information is *very* easily changed. Billing
should be based on BTN and routing should be based on ANI, originating
trunk group or other trunk specific information (ie: for proper routing
of 911 calls).
By the way, if anyone out there would like to comment on that premise,
I'd love to hear what some of you think regarding routing/rating via
callerID data.
Now if you do use ANI, or BTN for this lookup, you will have to query
NPAC to verify who actually owns the number since it could have been
ported. I would never trust a LERG 6 lookup to tell me the originating
carrier.
Also, I don't think you can query LIDB for originating carrier
information. LIDB (Line Information DataBase) is primarily used to store
line options for things like if you allow collect calls, 3rd party
billing, what kind of line it is (business/res), calling card numbers (I
believe this is an antiquated use of the system), and sometimes the CNAM
(caller name) database is coupled with LIDB (as it is with Verisign). In
fact, I have access to LIDB and it wouldn't let me touch (query or
otherwise) anything that isn't me.
-Brett
Matt Klein wrote:
More specifically, see the data sheet about lidb:
http://www.verisign.com/stellent/groups/public/documents/data_sheet/001944.pdf
You could go that route, or get a switch, or... there's a variety of other
options. But if you're looking for a full number lookup, you're looking
for lidb access..
-m
On Sun, 26 Dec 2004, Lyle Giese wrote:
That's good to get a general idea, but number portability only tells you
which carrier has the block. It does not let you know about specific
numbers :-{
Lyle
- Original Message -
From: Matt Klein [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, December 26, 2004 5:11 PM
Subject: Re: [Asterisk-Users] OT - Originating Network identity
focus on npa-nxx (area code-prefix)
if the call is coming from a non-ported number, then
http://telcodata.us/docs/queries.html may help you --
see the example files..
there are also a couple other sites out there.. but i've
found this one to be my favorite thus far.
-m
On Sun, 26 Dec 2004, oi geli wrote:
I am not sure if it is the right list for the post.
Please excuse my lack of expertise, if it is a bad
post.
Is there anyway to detect the originating network
identity of the call in Asterisk? For example, if the
Asterisk gets a call from Cingular Network, is there
anyway to find out that the call came from a Cingular
subscriber.
Thanks
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[Asterisk-Users] RINGBACK then HANGUP
I am using the manager API to sucessfully ORIGINATE a call. I am using PHP.
I connect to asterisk and then connect an internal SIP phone to an external
phone.
?php
$timeout = 7500;
$login_extension = SIP/6001; // agent extension
$call_telephone = 9707; // customer's telephone
$socket = fsockopen(10.0.0.3,5038, $errno, $errstr, $timeout);
if ($socket)
{
$call_person = Exten: . $call_telephone . \r\n;
$call_extension = Channel: . $login_extension . \r\n;
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: user\r\n);
fputs($socket, Secret: password\r\n\r\n);
fputs($socket, Action: Originate\r\n);
fputs($socket, $call_extension);
fputs($socket, Context: local\r\n);
fputs($socket, $call_person);
fputs($socket, Priority: 1\r\n);
fputs($socket, Callerid: \r\n\r\n);
}
?
The above code is used when I need to ringback a customer to tell them
their driver is outside. Problem is the call centre agent initiates the
call. The agent's SIP phone (SIP/6001) rings then he answers the call from
asterisk, asterisk then dials the customer.
I want asterisk to dial the customer, ring twice and then hangup. This will
save the agent's time and reduce our call costs. I don't want the agent to
be involved. I have tried messing around with the code above but no result.
So my problem in summary is: I would like dial an external line, let it ring
twice and then hangup all via PHP.
Thanks for your help.
Gary Ruddock
swiftdrinks
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[Asterisk-Users] Supporting End User Line Features
Sigh.. This shouldn't be so hard. Ok guys, I'm trying to figure out how to support end user features for my users. Perhaps some of them are typical verticle service features like *69, *72, *66, etc, you get the picture. Here's my deal. Sure implementing them one by one is easy enough. But building the logic on the incoming side (PSTN calling my SIP customer for example) is a real pain: 1. Call comes in from pstn 2. is destination allowed to call forward? 3. If call forwarding is allowed, is it enabled? 4. If it's enabled, what's the number? 5. Is do not disturbed allowed/enabled? 6. Is call return allowed/enabled? 7. if call return is enabled, store the incoming callerid 8. etc. etc. etc So a standard extension could end up being really really really long to support all these features.. Just seems so.. wrong considering that some of my customers might have no features at all. I seem to remember that Zoa mentioned that gotos were horribly slow and I'm planning on really loading these machines up with simultainious (100% G711 SIP) calls. So does anyone have any ideas on some simple logic that doesn't require each and every call to go through all these steps? I can't seem to think of a way.. Other than doing funky pattern matches... Then again, if the gotos arn't such a big deal, or if having 40-80 actual steps before a call is sent to the phone is ok, hey, I'm ok with that. Anyone have any experience with that? It just.. feels so wrong.. Seems like a gosub might be what I need shrug Any thoughts? Heh, I know chances are that the solution is probably a lot more obvious than I'm making it. So if it is, please be gentle with me. :) Thanks, -Brett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending call to analog then to Vmailaftertimeout?
[macro-stdcs]
;
;; Call a device with cs ;;
;; Takes 2 arguments ;;
;; arg1 exten ;;
;; arg2 device;;
;; tnen goes to vm ;;
;
;screen-record: Please record your name press pound when finished.
;screen-from: You have a call from
;screen-accept: Press 1 to accept 2 to reject, and 3 to transfer.
exten = s,1,Wait(0.2)
exten = s,2,Playback(vm-rec-name)
exten = s,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
exten = s,4,Record(${SCREEN_FILE}.gsm|2|4)
exten = s,5,Playback(pls-wait-connect-call)
exten = s,6,Dial(${ARG2},30,mtM(screen^${SCREEN_FILE}))
exten = s,7,Goto(17);VM
'I always leaeve room for more in case the dial plan changes
exten = s,17,Voicemail(u${ARG1})
exten = s,18,Playback(goodbye)
exten = s,19,Hangup
exten = s,107,Goto(17)
exten = h,1,System(/bin/rm ${ARG1}.gsm)
[macro-screen]
;this is called in the Dial statement using M
;ARG1 recorded name to play back
;TODO: add a response timeout, after which the message is repeated
(needed for outgoing zap fxo channels) and absolute timeout, after
which VM is used
exten = s,1,noop(${ARG1})
exten = s,2,Playback(custom/screen-from) ;you have an incoming call from:
exten = s,3,Playback(${ARG1})
;press 1 to accept 2 to reject 3 to transfer
exten = s,4,Read(ACCEPT|custom/screnn-accept|1)
exten = s,5,Gotoif($[${ACCEPT} = 1] ?50) ;connect
exten = s,6,Gotoif($[${ACCEPT} = 2] ?30) ;reject to vm
exten = s,7,Gotoif($[${ACCEPT} = 3] ?40) ;TRANSFER
exten = s,8,Gotoif($[${ACCEPT} = 4] ?30:30) ;any thing else vm
exten = s,30,SetVar(MACRO_RESULT=CONTINUE)
exten = s,31,Goto(50)
exten = s,40,Read(TEXTEN|custom/screen-exten|3)
;ask for extension then set macro to goto that and continue
exten = s,41,Gotoif($[${LEN(${TEXTEN})} = 3]?42:45)
exten = s,42,SetVar(MACRO_RESULT=GOTO:internaldial^${TEXTEN}^1)
exten = s,43,Goto(50)
exten = s,45,Gotoif($[${TEXTEN} = 0] ?46:46)
;the logic is here to allow transfer to operator, i just didn't imlepent it yet
exten = s,46,SetVar(MACRO_RESULT=CONTINUE)
exten = s,47,Goto(50)
exten = s,50,System(/bin/rm ${ARG1}.gsm)
exten = h,1,System(/bin/rm ${ARG1}.gsm)
On Wed, 29 Dec 2004 00:35:34 -0600, Me [EMAIL PROTECTED] wrote:
Nevermind, it looks like Asterisk cmd Read is my lucky command :)
Thanks!
Start Your Own Internet Service!
http://www.YourOwnISP.com
- Original Message -
From: Me [EMAIL PROTECTED]
To: C F [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Wednesday, December 29, 2004 12:19 AM
Subject: Re: [Asterisk-Users] Sending call to analog then to
Vmailaftertimeout?
I was trying this logic before, I got as far as going into the Macro,
playing a message and then.. Well... I got lost, I am not sure how to go
about require them to press a button. Normally I can make someone press an
extension but from what I read about Macros in * you have to stay within
the
s extension.
Any idea where I can find an example of this sort of thing?
Thanks!
Start Your Own Internet Service!
http://www.YourOwnISP.com
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 28, 2004 11:34 PM
Subject: Re: [Asterisk-Users] Sending call to analog then to
Vmailaftertimeout?
-- Forwarded message --
From: C F [EMAIL PROTECTED]
Date: Wed, 29 Dec 2004 00:34:28 -0500
Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail
aftertimeout?
To: Me [EMAIL PROTECTED]
try the M option which will do a macro and will not connect the caller
unless s/he presses some button. and if no button is pressed then it
goes to VM. now remember to replay the message (to press the button) a
few times b4 going to VM otherwise they will never hear it, since *
considers it answered .
http://www.voip-info.org/wiki-Asterisk+cmd+dial
On Tue, 28 Dec 2004 23:29:54 -0600, Me [EMAIL PROTECTED] wrote:
I was aware of the c option but it's a pain for people to have to
press
the # sign plus they have to know they are suppose to do that. In
addition,
I tried to use the A option to play a sound to them when they answer
reminding them to press pound at the end of the message but the sound
doesn't play until they press pound :)
So.. It appears I am still stuck with * considering the call answered
when
the Zap channels grabs it and connects the other leg of the call.
Hopefully
there is some other way to make this happen.
Thanks for the feedback though.
Start Your Own Internet Service!
http://www.YourOwnISP.com
[Asterisk-Users] IP Phone recommendations?
Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have orphans around. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I was hoping some folks would share what they have found. My primary goal is to replicate the 7310's features and to allow room for growth in the future with telephony applications. Our primary driver is configurability and features that we can get in Asterisk, that we can get without a lot of money from Nortel. Namely- Voicemail, telecommuting workers on the pbx, better call handling, better automation. I'd like to be able to integrate smart features like directory and call handling to the handset, but I'll freely admit I'm just starting out. My initial goal is to just to get onto Asterisk and get it working. I'll worry about cool stuff later. Our integration and migration plan is as follows: If anyone has some suggestions or pointers I'd love to hear them. 1. Test and evaluate Asterisk with TDM400 with 1 FXO/FSO port each. 2. Configure Asterisk to be the primary PBX and slave the Nortel Meridian system to it using a second TDM400. This avoids immediate replacement of all handsets. Will allow immediate access to features such as Voicemail. 3. Overtime, upgrade desk phones to IP phones. When all phones are replaced, decommission Nortel and sell on Ebay. :) Cold turkey option is to spend the extra $ and buy the handsets upfront and just ditch nortel without a transition period. We currently have 4 pbx lines and 1 dedicated fax/credit card line. We have 10 handsets. Thanks, Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
So we could provide caller position announcements without the callers actually incurring charges? Has anybody tried this (in the UK)? -Original Message- From: Peter Svensson [mailto:[EMAIL PROTECTED] Sent: 29 December 2004 14:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Music instead of Tunes On Wed, 29 Dec 2004, Steve Hanselman wrote: Are the two cases different in any way? The external call comes in, goes to a context which eventually leads to a Dial(...) calling the internal user. That Dial call provides music to the external caller while the internal call is in progress. The difference is that you'd have to answer the call, my guess is that it can't be done (by a Joe Average like ourselves), otherwise we'd provide useful information to callers at no charge. For pots lines this is true. For isdn lines there is no need to answer prior to sending data. The reverse path (from the called party towards the calling party) is opened when (this is form memory, it may be another IE) PROGRESS is transmitted. You can use Playback and a host of other connads on an unanswered line. Some of these will automatically answer the line unless given an option not to. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX - IAX - SIP problems
The setup: Inc SIP Call - Asterisk 1 -- IAX -- Asterisk 2 -- SIP -- phone (3044) Asterisk 1 shows the following: (1.0.3) -- Executing Goto(SIP/XX.XX.XX.XX-0819f590, cytel-internal|3044|1) in new stack -- Goto (cytel-internal,3044,1) -- Executing Dial(SIP/XX.XX.XX.XX-0819f590, IAX2/asterisk-alpha:[EMAIL PROTECTED]/3044|30) in new stack -- Called asterisk-alpha:[EMAIL PROTECTED]/3044 -- Call accepted by XX.XX.XX.XX (format ulaw) -- Format for call is ulaw -- Hungup 'IAX2/devasterisk/1' Asterisk 2 shows the following: (CVS-HEAD) -- Registered SIP '3044' at XX.XX.XX.XX port 1911 expires 3600 -- Saved useragent CSCO/7 for peer 3044 -- Accepting AUTHENTICATED call from XX.XX.XX.XX, requested format = 4, actual format = 4 -- Executing Dial(IAX2/[EMAIL PROTECTED]/2, SIP/3044,30) Dec 29 08:36:09 WARNING[1496]: chan_sip.c:1351 create_addr: No such host: 3044,30 Dec 29 08:36:09 NOTICE[1496]: app_dial.c:803 dial_exec: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) -- Hungup 'IAX2/[EMAIL PROTECTED]/2' So how can Asterisk2 say 'no such host' when it just registered 3044 30seconds before this call came in? I'm not using any RealTime stuff anywhere. 3044 is defined: [3044] md5secret=d2756499745e254f52a224713f1a7d91 type=friend host=dynamic nat=yes canreinvite=yes disallow=g729 context=cytel-internal any ideas? Thanks, Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending call to analog then to Vmailaftertimeout?
Thanks for the example! I was using something similar to this that I found
in the Wiki but the problem I ran into was the Record() part. Each time *
got to the record part I got some error saying, can't remember what it was,
I will dig it up and post it in a reply.
Start Your Own Internet Service!
http://www.YourOwnISP.com
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, December 29, 2004 9:41 AM
Subject: Re: [Asterisk-Users] Sending call to analog then to
Vmailaftertimeout?
[macro-stdcs]
;
;; Call a device with cs;;
;; Takes 2 arguments ;;
;; arg1 exten ;;
;; arg2 device ;;
;; tnen goes to vm;;
;
;screen-record: Please record your name press pound when finished.
;screen-from: You have a call from
;screen-accept: Press 1 to accept 2 to reject, and 3 to transfer.
exten = s,1,Wait(0.2)
exten = s,2,Playback(vm-rec-name)
exten = s,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
exten = s,4,Record(${SCREEN_FILE}.gsm|2|4)
exten = s,5,Playback(pls-wait-connect-call)
exten = s,6,Dial(${ARG2},30,mtM(screen^${SCREEN_FILE}))
exten = s,7,Goto(17);VM
'I always leaeve room for more in case the dial plan changes
exten = s,17,Voicemail(u${ARG1})
exten = s,18,Playback(goodbye)
exten = s,19,Hangup
exten = s,107,Goto(17)
exten = h,1,System(/bin/rm ${ARG1}.gsm)
[macro-screen]
;this is called in the Dial statement using M
;ARG1 recorded name to play back
;TODO: add a response timeout, after which the message is repeated
(needed for outgoing zap fxo channels) and absolute timeout, after
which VM is used
exten = s,1,noop(${ARG1})
exten = s,2,Playback(custom/screen-from) ;you have an incoming call from:
exten = s,3,Playback(${ARG1})
;press 1 to accept 2 to reject 3 to transfer
exten = s,4,Read(ACCEPT|custom/screnn-accept|1)
exten = s,5,Gotoif($[${ACCEPT} = 1] ?50) ;connect
exten = s,6,Gotoif($[${ACCEPT} = 2] ?30) ;reject to vm
exten = s,7,Gotoif($[${ACCEPT} = 3] ?40) ;TRANSFER
exten = s,8,Gotoif($[${ACCEPT} = 4] ?30:30) ;any thing else vm
exten = s,30,SetVar(MACRO_RESULT=CONTINUE)
exten = s,31,Goto(50)
exten = s,40,Read(TEXTEN|custom/screen-exten|3)
;ask for extension then set macro to goto that and continue
exten = s,41,Gotoif($[${LEN(${TEXTEN})} = 3]?42:45)
exten = s,42,SetVar(MACRO_RESULT=GOTO:internaldial^${TEXTEN}^1)
exten = s,43,Goto(50)
exten = s,45,Gotoif($[${TEXTEN} = 0] ?46:46)
;the logic is here to allow transfer to operator, i just didn't imlepent
it yet
exten = s,46,SetVar(MACRO_RESULT=CONTINUE)
exten = s,47,Goto(50)
exten = s,50,System(/bin/rm ${ARG1}.gsm)
exten = h,1,System(/bin/rm ${ARG1}.gsm)
On Wed, 29 Dec 2004 00:35:34 -0600, Me [EMAIL PROTECTED] wrote:
Nevermind, it looks like Asterisk cmd Read is my lucky command :)
Thanks!
Start Your Own Internet Service!
http://www.YourOwnISP.com
- Original Message -
From: Me [EMAIL PROTECTED]
To: C F [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Wednesday, December 29, 2004 12:19 AM
Subject: Re: [Asterisk-Users] Sending call to analog then to
Vmailaftertimeout?
I was trying this logic before, I got as far as going into the Macro,
playing a message and then.. Well... I got lost, I am not sure how to
go
about require them to press a button. Normally I can make someone
press an
extension but from what I read about Macros in * you have to stay
within
the
s extension.
Any idea where I can find an example of this sort of thing?
Thanks!
Start Your Own Internet Service!
http://www.YourOwnISP.com
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 28, 2004 11:34 PM
Subject: Re: [Asterisk-Users] Sending call to analog then to
Vmailaftertimeout?
-- Forwarded message --
From: C F [EMAIL PROTECTED]
Date: Wed, 29 Dec 2004 00:34:28 -0500
Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail
aftertimeout?
To: Me [EMAIL PROTECTED]
try the M option which will do a macro and will not connect the
caller
unless s/he presses some button. and if no button is pressed then it
goes to VM. now remember to replay the message (to press the button)
a
few times b4 going to VM otherwise they will never hear it, since *
considers it answered .
http://www.voip-info.org/wiki-Asterisk+cmd+dial
On Tue, 28 Dec 2004 23:29:54 -0600, Me [EMAIL PROTECTED]
wrote:
I was aware of the c option but it's a pain for people to have
to
press
the # sign plus they have to know they are suppose to
[Asterisk-Users] Mysql-Realtime
Hi ALL; Hi matthew; Thanks to mark and matthew for creation of Real-time, It works ok for me.Since Im not an expert in coding, I come up with the following questions: 1) As I found, the real-time driver is inasterisk-addons.Where is the specific code about real-time inside the asterisk? 2) Can we change the Queries and fields? Warmest Regads Mohammad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IP Phone recommendations?
Many use cisco IP phones, almost any model. Support and firmware access has a fee. SNOM 190 works well, free firmware, good community support. Lots of reports of good luck with Polycom phones (IP500), but they wont provide any support when used with * and you have to get your firmware from the net, not from polycom, even if are willing to pay. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 8:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IP Phone recommendations? Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have orphans around. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I was hoping some folks would share what they have found. My primary goal is to replicate the 7310's features and to allow room for growth in the future with telephony applications. Our primary driver is configurability and features that we can get in Asterisk, that we can get without a lot of money from Nortel. Namely- Voicemail, telecommuting workers on the pbx, better call handling, better automation. I'd like to be able to integrate smart features like directory and call handling to the handset, but I'll freely admit I'm just starting out. My initial goal is to just to get onto Asterisk and get it working. I'll worry about cool stuff later. Our integration and migration plan is as follows: If anyone has some suggestions or pointers I'd love to hear them. 1. Test and evaluate Asterisk with TDM400 with 1 FXO/FSO port each. 2. Configure Asterisk to be the primary PBX and slave the Nortel Meridian system to it using a second TDM400. This avoids immediate replacement of all handsets. Will allow immediate access to features such as Voicemail. 3. Overtime, upgrade desk phones to IP phones. When all phones are replaced, decommission Nortel and sell on Ebay. :) Cold turkey option is to spend the extra $ and buy the handsets upfront and just ditch nortel without a transition period. We currently have 4 pbx lines and 1 dedicated fax/credit card line. We have 10 handsets. Thanks, Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] service activation code
Hi; Can we have "Activation Codes" on Sip phones? Regards Mohammad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IP Phone recommendations?
Okay, I'm feeling a little stupid here But I'm gonna ask anyway. You mention support and firmware on the Ci$co phones. I understand the support item. I guess it makes sense that the phones have firmware. Does it have to be updated or changed or messed with that often? If there is an article somewhere that covers this I'd love to read it. It seems like most of the VOIP marketing-speak is aimed at companies with mega$$$ who want to spend $500/head on it. We're a tad smaller and we have $ to spend not $$ or $$$ or . :) Worse yet, we need $ to go find and bring back it's friends. :) Anyhow, I haven't seen anything that really tackles moving from a CISC Nortel Meridian KSU to a IP based system. I'm guessing that this is Nortel's absolute worst nightmare. It seems like they trickle down the technology from the large switches to the micro PBX systems. Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: Damon Estep [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 10:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IP Phone recommendations? Many use cisco IP phones, almost any model. Support and firmware access has a fee. SNOM 190 works well, free firmware, good community support. Lots of reports of good luck with Polycom phones (IP500), but they wont provide any support when used with * and you have to get your firmware from the net, not from polycom, even if are willing to pay. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 8:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IP Phone recommendations? Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have orphans around. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I was hoping some folks would share what they have found. My primary goal is to replicate the 7310's features and to allow room for growth in the future with telephony applications. Our primary driver is configurability and features that we can get in Asterisk, that we can get without a lot of money from Nortel. Namely- Voicemail, telecommuting workers on the pbx, better call handling, better automation. I'd like to be able to integrate smart features like directory and call handling to the handset, but I'll freely admit I'm just starting out. My initial goal is to just to get onto Asterisk and get it working. I'll worry about cool stuff later. Our integration and migration plan is as follows: If anyone has some suggestions or pointers I'd love to hear them. 1. Test and evaluate Asterisk with TDM400 with 1 FXO/FSO port each. 2. Configure Asterisk to be the primary PBX and slave the Nortel Meridian system to it using a second TDM400. This avoids immediate replacement of all handsets. Will allow immediate access to features such as Voicemail. 3. Overtime, upgrade desk phones to IP phones. When all phones are replaced, decommission Nortel and sell on Ebay. :) Cold turkey option is to spend the extra $ and buy the handsets upfront and just ditch nortel without a transition period. We currently have 4 pbx lines and 1 dedicated fax/credit card line. We have 10 handsets. Thanks, Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phone recommendations?
Why not use ATA adapters? This way you can use just about any phone you want. Start Your Own Internet Service! http://www.YourOwnISP.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, December 29, 2004 10:28 AM Subject: RE: [Asterisk-Users] IP Phone recommendations? Okay, I'm feeling a little stupid here But I'm gonna ask anyway. You mention support and firmware on the Ci$co phones. I understand the support item. I guess it makes sense that the phones have firmware. Does it have to be updated or changed or messed with that often? If there is an article somewhere that covers this I'd love to read it. It seems like most of the VOIP marketing-speak is aimed at companies with mega$$$ who want to spend $500/head on it. We're a tad smaller and we have $ to spend not $$ or $$$ or . :) Worse yet, we need $ to go find and bring back it's friends. :) Anyhow, I haven't seen anything that really tackles moving from a CISC Nortel Meridian KSU to a IP based system. I'm guessing that this is Nortel's absolute worst nightmare. It seems like they trickle down the technology from the large switches to the micro PBX systems. Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: Damon Estep [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 10:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IP Phone recommendations? Many use cisco IP phones, almost any model. Support and firmware access has a fee. SNOM 190 works well, free firmware, good community support. Lots of reports of good luck with Polycom phones (IP500), but they wont provide any support when used with * and you have to get your firmware from the net, not from polycom, even if are willing to pay. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 8:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IP Phone recommendations? Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have orphans around. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I was hoping some folks would share what they have found. My primary goal is to replicate the 7310's features and to allow room for growth in the future with telephony applications. Our primary driver is configurability and features that we can get in Asterisk, that we can get without a lot of money from Nortel. Namely- Voicemail, telecommuting workers on the pbx, better call handling, better automation. I'd like to be able to integrate smart features like directory and call handling to the handset, but I'll freely admit I'm just starting out. My initial goal is to just to get onto Asterisk and get it working. I'll worry about cool stuff later. Our integration and migration plan is as follows: If anyone has some suggestions or pointers I'd love to hear them. 1. Test and evaluate Asterisk with TDM400 with 1 FXO/FSO port each. 2. Configure Asterisk to be the primary PBX and slave the Nortel Meridian system to it using a second TDM400. This avoids immediate replacement of all handsets. Will allow immediate access to features such as Voicemail. 3. Overtime, upgrade desk phones to IP phones. When all phones are replaced, decommission Nortel and sell on Ebay. :) Cold turkey option is to spend the extra $ and buy the handsets upfront and just ditch nortel without a transition period. We currently have 4 pbx lines and 1 dedicated fax/credit card line. We have 10 handsets. Thanks, Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To
RE: [Asterisk-Users] IP Phone recommendations?
You could, but in a business environment the features of the phones are very useful, such as multiple call appearances without annoying call waiting beeps, a multi line display, a web interface for address books, remote firmware updates, the list goes on. ATAs are great for residential and fax machines (assuming the ATA is fax aware). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Me Sent: Wednesday, December 29, 2004 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IP Phone recommendations? Why not use ATA adapters? This way you can use just about any phone you want. Start Your Own Internet Service! http://www.YourOwnISP.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, December 29, 2004 10:28 AM Subject: RE: [Asterisk-Users] IP Phone recommendations? Okay, I'm feeling a little stupid here But I'm gonna ask anyway. You mention support and firmware on the Ci$co phones. I understand the support item. I guess it makes sense that the phones have firmware. Does it have to be updated or changed or messed with that often? If there is an article somewhere that covers this I'd love to read it. It seems like most of the VOIP marketing-speak is aimed at companies with mega$$$ who want to spend $500/head on it. We're a tad smaller and we have $ to spend not $$ or $$$ or . :) Worse yet, we need $ to go find and bring back it's friends. :) Anyhow, I haven't seen anything that really tackles moving from a CISC Nortel Meridian KSU to a IP based system. I'm guessing that this is Nortel's absolute worst nightmare. It seems like they trickle down the technology from the large switches to the micro PBX systems. Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: Damon Estep [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 10:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IP Phone recommendations? Many use cisco IP phones, almost any model. Support and firmware access has a fee. SNOM 190 works well, free firmware, good community support. Lots of reports of good luck with Polycom phones (IP500), but they wont provide any support when used with * and you have to get your firmware from the net, not from polycom, even if are willing to pay. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 8:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IP Phone recommendations? Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have orphans around. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I was hoping some folks would share what they have found. My primary goal is to replicate the 7310's features and to allow room for growth in the future with telephony applications. Our primary driver is configurability and features that we can get in Asterisk, that we can get without a lot of money from Nortel. Namely- Voicemail, telecommuting workers on the pbx, better call handling, better automation. I'd like to be able to integrate smart features like directory and call handling to the handset, but I'll freely admit I'm just starting out. My initial goal is to just to get onto Asterisk and get it working. I'll worry about cool stuff later. Our integration and migration plan is as follows: If anyone has some suggestions or pointers I'd love to hear them. 1. Test and evaluate Asterisk with TDM400 with 1 FXO/FSO port each. 2. Configure Asterisk to be the primary PBX and slave the Nortel Meridian system to it using a second TDM400. This avoids immediate replacement of all handsets. Will allow immediate access to features such as Voicemail. 3. Overtime, upgrade desk phones to IP phones. When all phones are replaced, decommission Nortel and sell on Ebay. :) Cold turkey option is to spend the extra $ and buy the handsets upfront and just ditch nortel without a transition period. We currently have 4 pbx lines and 1 dedicated fax/credit card line. We have 10 handsets. Thanks, Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
RE: [Asterisk-Users] IP Phone recommendations?
You will always want access to firmware, SIP phones evolve quickly and new features and bug fixes are usually implanted via firmware updates. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 9:29 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] IP Phone recommendations? Okay, I'm feeling a little stupid here But I'm gonna ask anyway. You mention support and firmware on the Ci$co phones. I understand the support item. I guess it makes sense that the phones have firmware. Does it have to be updated or changed or messed with that often? If there is an article somewhere that covers this I'd love to read it. It seems like most of the VOIP marketing-speak is aimed at companies with mega$$$ who want to spend $500/head on it. We're a tad smaller and we have $ to spend not $$ or $$$ or . :) Worse yet, we need $ to go find and bring back it's friends. :) Anyhow, I haven't seen anything that really tackles moving from a CISC Nortel Meridian KSU to a IP based system. I'm guessing that this is Nortel's absolute worst nightmare. It seems like they trickle down the technology from the large switches to the micro PBX systems. Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: Damon Estep [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 10:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IP Phone recommendations? Many use cisco IP phones, almost any model. Support and firmware access has a fee. SNOM 190 works well, free firmware, good community support. Lots of reports of good luck with Polycom phones (IP500), but they wont provide any support when used with * and you have to get your firmware from the net, not from polycom, even if are willing to pay. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 8:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IP Phone recommendations? Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have orphans around. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I was hoping some folks would share what they have found. My primary goal is to replicate the 7310's features and to allow room for growth in the future with telephony applications. Our primary driver is configurability and features that we can get in Asterisk, that we can get without a lot of money from Nortel. Namely- Voicemail, telecommuting workers on the pbx, better call handling, better automation. I'd like to be able to integrate smart features like directory and call handling to the handset, but I'll freely admit I'm just starting out. My initial goal is to just to get onto Asterisk and get it working. I'll worry about cool stuff later. Our integration and migration plan is as follows: If anyone has some suggestions or pointers I'd love to hear them. 1. Test and evaluate Asterisk with TDM400 with 1 FXO/FSO port each. 2. Configure Asterisk to be the primary PBX and slave the Nortel Meridian system to it using a second TDM400. This avoids immediate replacement of all handsets. Will allow immediate access to features such as Voicemail. 3. Overtime, upgrade desk phones to IP phones. When all phones are replaced, decommission Nortel and sell on Ebay. :) Cold turkey option is to spend the extra $ and buy the handsets upfront and just ditch nortel without a transition period. We currently have 4 pbx lines and 1 dedicated fax/credit card line. We have 10 handsets. Thanks, Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update
Re: [Asterisk-Users] Mysql-Realtime
Thanks to mark and matthew for creation of Real-time Holy crap. I didn't create RealTime (just the MySQL driver). RealTime iteself is all on Mark and Anthm. 1) As I found, the real-time driver is in asterisk-addons. Correction..the MySQL RealTime driver is in asterisk-addons. The ODBC RealTime driver is built-in Where is the specific code about real-time inside the asterisk? What do you mean 'specific code'? The majority of the code for RealTime lies in the drivers. Thats the beauty of RealTime. There is some code that handles the registration and execution of the drivers in config.c. 2) Can we change the Queries and fields ? You can change the queries to an extent; moreso now than before thanks to Marks changes. When you issue an ast_realtime_load on a specific family, you get back (AFAIK) all fields in the table that are not NULL. Look at chan_sip, chan_iax, pbx_realtime for sample API code. Or just grep the asterisk source tree for ast_load_realtime -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IP Phone recommendations?
Ah That makes sense. Thanks for pointing that out. I'm so used to thinking of phone as bricks with a wire and handset and some stress relief buttons. :) Your other post about the ATA adapters is dead-on. Multi-line display and friends (address book etc) are invaluable. I thought about going soft-phone. But I'm not sure I trust our windows systems that much. We still have a couple of applications that have us hogtied to windows. Evil UPS OnlineWorldship is one such poorly written application. Ironic that UPS uses Unix on the backend but won't support it for clients. I have about 8 Win9x machines because of that issue. I'm eyeing a flight from Exchange/Outlook later this year. And our graphics stuff is all Windows/Adobe based. I guess the thing with soft-phones is that I'm not sure how stable the machines are. It's bad enough to lose your phone or your computer, but both at once is really spooky. ;) On a bright note, we have gigabit everywhere in the building. So QOS and bandwidth had better not be an issue internally. :) Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: Damon Estep [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IP Phone recommendations? You will always want access to firmware, SIP phones evolve quickly and new features and bug fixes are usually implanted via firmware updates. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 9:29 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] IP Phone recommendations? Okay, I'm feeling a little stupid here But I'm gonna ask anyway. You mention support and firmware on the Ci$co phones. I understand the support item. I guess it makes sense that the phones have firmware. Does it have to be updated or changed or messed with that often? If there is an article somewhere that covers this I'd love to read it. It seems like most of the VOIP marketing-speak is aimed at companies with mega$$$ who want to spend $500/head on it. We're a tad smaller and we have $ to spend not $$ or $$$ or . :) Worse yet, we need $ to go find and bring back it's friends. :) Anyhow, I haven't seen anything that really tackles moving from a CISC Nortel Meridian KSU to a IP based system. I'm guessing that this is Nortel's absolute worst nightmare. It seems like they trickle down the technology from the large switches to the micro PBX systems. Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: Damon Estep [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 10:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IP Phone recommendations? Many use cisco IP phones, almost any model. Support and firmware access has a fee. SNOM 190 works well, free firmware, good community support. Lots of reports of good luck with Polycom phones (IP500), but they wont provide any support when used with * and you have to get your firmware from the net, not from polycom, even if are willing to pay. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 8:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IP Phone recommendations? Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have orphans around. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I was hoping some folks would share what they have found. My primary goal is to replicate the 7310's features and to allow room for growth in the future with telephony applications. Our primary driver is configurability and features that we can get in Asterisk, that we can get without a lot of money from Nortel. Namely- Voicemail, telecommuting workers on the pbx, better call handling, better automation. I'd like to be able to integrate smart features like directory and call handling to the handset, but I'll freely admit I'm just starting out. My initial goal is to just to get onto Asterisk and get it working. I'll worry about cool stuff later. Our integration and migration plan is as follows: If anyone has some suggestions or pointers I'd love to hear them. 1. Test and evaluate Asterisk with TDM400 with 1 FXO/FSO port each. 2. Configure Asterisk to be the primary PBX and slave the Nortel Meridian
Re: [Asterisk-Users] IP Phone recommendations?
ATAs are great for residential Linksys PAP2-NA is a 2 line ATA for $50 resale. Works great with Asterisk. and fax machines (assuming the ATA is fax aware). Just because your ATA is fax aware doesn't guarantee it will work. We use the PAP2-NA's and they are fax aware but don't work as reliabaly as they should. Hopefully Asterisk will come out with T.38 support for FoIP. There are some ATA's out there that support fax detection and use T.38 but since Asterisk doesn't support T.38, the fax probably won't go thru. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phone recommendations?
[EMAIL PROTECTED] wrote: Ah That makes sense. Thanks for pointing that out. I'm so used to thinking of phone as bricks with a wire and handset and some stress relief buttons. :) Your other post about the ATA adapters is dead-on. Multi-line display and friends (address book etc) are invaluable. I thought about going soft-phone. But I'm not sure I trust our windows systems that much. We still have a couple of applications that have us hogtied to windows. Evil UPS OnlineWorldship is one such poorly written application. Ironic that UPS uses Unix on the backend but won't support it for clients. I have about 8 Win9x machines because of that issue. I'm eyeing a flight from Exchange/Outlook later this year. And our graphics stuff is all Windows/Adobe based. I guess the thing with soft-phones is that I'm not sure how stable the machines are. It's bad enough to lose your phone or your computer, but both at once is really spooky. ;) On a bright note, we have gigabit everywhere in the building. So QOS and bandwidth had better not be an issue internally. :) Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) Brian, You really, really want to get SIP phones. Softphones (Windows, PC hardware) just do not have the quality and reliability that dedicated hardware SIP phones do. If your windows computers go down with the latest worm, you won't even be able to call someone to ask for help or answer the phone when customers start complaining. It's really pretty obvious here. Also, I am a big Polycom fan. The three phones that you would look at are very rich in features, and range from $115 - $255. Getting one brand really makes things easier, and Cisco sure doesn't offer anything for $115. The 7960 does not have one feature (I know of) that the Polycom IP 600 ($255) does not have. The Polycom's work better for paging, intercom, presence, conferencing, and more. The 7960's seem to be running off of brand recognition. Don't get me wrong, they are excellent phones, but I feel that they just don't compete with the Polycom offering in features. Think of it this way: Polycom is like Avis - They try harder. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXy echo...
I have set up asterisk with 3 or 4 phones at home, trying to create a development lab of sorts. I subscribe to the my wife is my guinea pig philosophy. What I have noticed is that on my IAXy I get a lot of echo, but it is spuratic at best. I have not experienced any echo on the other phones (cisco 7960, polycom IP500, SNOM 220). My PSTN connection is through a X100P from digium and I have echo cancel on. Has anyone else encountered this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Perhaps something obvious?
I am a VoicePulse.com user although I have never been able to connect. I have no dialtone nor can I determine if I have been authenticated. Do I need to configure for sip? I was told I did not need SIP. Voicepulse does support sip . . . Asterisk does start and runs stably. I can login locally with asterisk -r no problem. When I logged into my SPA 2000 using its web interface I noticed it is not registered. Below are my ipfilter and ipnat rules. (The firewall/ gateway is FreeBSD 5.3 but since I could not compile asterisk on FreeBSD 5.3 an internal gentoo machine is running it.) from /var/log/asterisk/messages: Dec 29 03:05:19 WARNING[18636]: Unable to open IAX timing interface: No such file or directory Dec 29 03:05:20 WARNING[18636]: Unable to get our IP address, Skinny disabled Dec 29 03:05:20 WARNING[18636]: Read error on sound device: Resource temporarily unavailable Dec 29 03:05:20 WARNING[18636]: Unable to get IP address for localhost.localdomain, SIP disabled Dec 29 03:07:48 WARNING[18664]: Unable to open pseudo channel for timing... Sound may be choppy. Dec 29 03:07:48 WARNING[18664]: Unable to get our IP address, MGCP disabled I am able to access the Internet in any other protocol from the Asterisk/Gentoo box. ipnat.conf: rdr fxp0 0.0.0.0/0 port 4569 - 10.0.0.147 port 4569 udp rdr fxp0 0.0.0.0/0 port 5036 - 10.0.0.147 port 5036 udp rdr fxp0 0.0.0.0/0 port 5060 - 10.0.0.147 port 5060 udp map fxp0 10.0.0.0/24 - 0/32 portmap tcp/udp 1:65000 map fxp0 10.0.0.0/24 - 0/32 pertinant ipf.conf rules: Internal NIC is vr0 pass in quick on vr0 from any to any pass out quick on vr0 from any to any External NIC is fxp0 but I need not mention it in the below rules. pass in quick proto udp from 66.234.228.170 to 24.98.219.30/32 port = 4569 group 10 pass in quick proto udp from 66.234.228.170 to 24.98.219.30/32 port = 5036 group 10 pass in quick proto udp from 66.234.228.170 to 24.98.219.30/32 port = 5060 group 10 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: RE: IP Phone recommendations?
-Original Message- Okay, I'm feeling a little stupid here But I'm gonna ask anyway. You mention support and firmware on the Ci$co phones. I understand the support item. I guess it makes sense that the phones have firmware. Does it have to be updated or changed or messed with that often? -because SIP UA's control the features (conference/transfer/redial etc) and each UA is under constant development to provide new features this will most likely be a requirement to keep current with the available features. If there is an article somewhere that covers this I'd love to read it. It seems like most of the VOIP marketing-speak is aimed at companies with mega$$$ who want to spend $500/head on it. We're a tad smaller and we have $ to spend not $$ or $$$ or . :) Worse yet, we need $ to go find and bring back it's friends. :) Anyhow, I haven't seen anything that really tackles moving from a CISC Nortel Meridian KSU to a IP based system. I'm guessing that this is Nortel's absolute worst nightmare. It seems like they trickle down the technology from the large switches to the micro PBX systems. - http://www.citel.com/index/index.asp I saw this yesterday and this may be an option for folks like you that have a big investment in handsets but want new features. I have no idea what one of these boxes costs but it looked interesting. Seems to be a play from Mitel to penetrate other mfg's installed bases with the 3300 ICP product. Remember that * is really a PBX where pretty much every system commercially available (other than class 5 equipment) is a hybrid of Key System technology and PBX functionality. Good Luck Jason Kawakami Open Telephony Labs, LLC Salt Lake City, UT ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel
On Wed, Dec 29, 2004 at 11:59:58AM +0100, Sergio Serrano wrote: Hi all, I have installed a TE110P in a BOX but when I load zaptel module I can't see any device in /proc/zaptel. And led of the card is green. From /proc/pci, it looks like you pci bus saw the card. Are you sure that you loaded the wcte11xp module? Matthew Fredrickson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: RE: IP Phone recommendations?
I agree with your comments about the KSU systems vs. * It just seems like every asian, latin american, and european electronics company has a VOIP product that they are selling now. Doesn't seem like we have seen any shakeout where a handful of companies emerge with good, solid products that perform day in and day out and provide a good value. I guess where we are really looking is for a SIP that is equivalent to our M7310 Nortel phones so that we can get rid of the Meridian system and it's phones. What I don't want is two PBX/KSU systems for 10 employees. NO WAY. :) I just figured that mentioning the M3710's might help clear the clutter of LG,Epson, Uniden, Emerson, Hampton Bay (kidding), etc aka crap at the bottom. I did consider ATA devices, but they cost half as much as a decent SIP appears to and they don't provide all the PBX features on their front. There is nothing worse then trying to teach employees secret phone codes to get to stuff. My impression is that the market shapes into three bands of phones: 1- Economy phones aimed at providing VOIP dial-tone substitutes. Under $100 2- Business Phones Between $100 and $200 3- Premium Business Phones From $250 My focus is squarely on 2 and possibly 3 if the phones are decent enough. NOTE: PLEASE DO NOT CALL AND SOLICIT ME. A couple of people have already tried and it's a real abuse of the listserv. I already own Asterisk, some digium equipment, and a production class server for it and I don't want to buy a box or a service provider's product. We prefer to be our service provider. Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: Jason Kawakami [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 11:23 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: RE: IP Phone recommendations? -Original Message- Okay, I'm feeling a little stupid here But I'm gonna ask anyway. You mention support and firmware on the Ci$co phones. I understand the support item. I guess it makes sense that the phones have firmware. Does it have to be updated or changed or messed with that often? -because SIP UA's control the features (conference/transfer/redial etc) and each UA is under constant development to provide new features this will most likely be a requirement to keep current with the available features. If there is an article somewhere that covers this I'd love to read it. It seems like most of the VOIP marketing-speak is aimed at companies with mega$$$ who want to spend $500/head on it. We're a tad smaller and we have $ to spend not $$ or $$$ or . :) Worse yet, we need $ to go find and bring back it's friends. :) Anyhow, I haven't seen anything that really tackles moving from a CISC Nortel Meridian KSU to a IP based system. I'm guessing that this is Nortel's absolute worst nightmare. It seems like they trickle down the technology from the large switches to the micro PBX systems. - http://www.citel.com/index/index.asp I saw this yesterday and this may be an option for folks like you that have a big investment in handsets but want new features. I have no idea what one of these boxes costs but it looked interesting. Seems to be a play from Mitel to penetrate other mfg's installed bases with the 3300 ICP product. Remember that * is really a PBX where pretty much every system commercially available (other than class 5 equipment) is a hybrid of Key System technology and PBX functionality. Good Luck Jason Kawakami Open Telephony Labs, LLC Salt Lake City, UT ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phone recommendations?
SIP may evolve quickly, but implementations do not IME. For a small operation I don't see this as a large issue. On Wed, 29 Dec 2004 09:39:06 -0700, Damon Estep [EMAIL PROTECTED] wrote: You will always want access to firmware, SIP phones evolve quickly and new features and bug fixes are usually implanted via firmware updates. -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Perhaps something obvious?
On Wed, Dec 29, 2004 at 12:21:51PM -0500, Matt Herzog wrote:
I am a VoicePulse.com user although I have never been able to connect.
I have no dialtone nor can I determine if I have been authenticated.
Do I need to configure for sip? I was told I did not need SIP.
Voicepulse does support sip . . .
And I forgot to ask, do I need to forward the ports to the Asterisk server
or the SPA device?
Here are my configs:
; Sample /etc/asterisk/iax.conf downloaded from VoicePulse and edited
; by MSH subsequently.
; Created September 1, 2004
[general]
port=5036
tos=lowdelay
jitterbuffer=no
; -
; The following codecs are support by the VoicePulse
; Connect! service:
; -
disallow=all
allow=ulaw
allow=ilbc
allow=gsm
allow=adpcm
allow=alaw
;allow=g726 ; g726 is NOT supported as of 10/1/2004,
; but is coming soon.
; -
; This is how you tell VoicePulse Connect! gateways where
; to send your incoming calls. The 10 characters before
; the : are your VoicePulse Connect! gateway login and
; the 10 characters after the colon are your Connect!
; gateway password. You can find this information by
; logging into your VoicePulse Connect! account at
; http://connect.voicepulse.com and clicking on Devices.
; -
;
; The entire register = line below should be on one line
; (with no carriage returns in the middle):
register = nkv87PBo43:[EMAIL PROTECTED]
register = nkv87PBo43:[EMAIL PROTECTED]
; -
; We use RSA keys for authentication purposes. If you
; haven't already saved the VoicePulse public key, you can
; get it by doing the following from a shell prompt:
;
; cd /var/lib/asterisk/keys
; wget http://connect.voicepulse.com/keys/voicepulse01.pub
; ( I installed their pub key. -- MSH )
; This is a guest user to catch all unauthenticated calls
;
[guest]
type=user
context=guest
;
; This is the VoicePulse user for incoming calls to your
; Asterisk server:
;
[voicepulse-in-01] ; -- Name must be [voicepulse-in-01]
type=user
context=incoming ; -- Should match the context you
auth=rsa
inkeys=voicepulse01
; This is a test user. You can use Dan Toma's DIAX Software
; Phone to test your Asterisk configuration. Set the
; following in the DIAX Config Registration menu option:
;
; Server: your Asterisk server IP address
; Username: diax
; Password: diaxpassword
;
; You can get DIAX at:
; http://www.laser.com/dante/diax/diax.html
; I eschew Windows.
;[diax]
;type=friend
;context=outgoing
;auth=md5
;secret=diaxpassword
;notransfer=1
;host=dynamic
;allow=gsm
; Sample /etc/asterisk/extensions.conf
; Created September 1, 2004
; Edited by MSH thereafter.
; =
; QUICKSTART WITH VOICEPULSE CONNECT! SERVICE:
; * Login to your VoicePulse Connect! account at:
; http://connect.voicepulse.com/
; * Go to the Devices tab and note your device login and
; password
; * Replace MY_DEVICE_LOGIN and MY_DEVICE_PASSWORD in the
; exten = statements below with your device login
; and password. (Lines 81-82)
; * If you DO NOT have a phone number from VoicePulse
; Connect!, comment out the following lines by placing a
; semicolon ; at the beginning:
; - The entire [arbitrary-name] context (lines 43-48)
; - The entire [testdtmf] context (lines 54-60)
; =
[general]
static=yes
writeprotect=no
[globals]
; [arbitrary-name] is the context referred to by the
; [voicepulse-in-01] user in iax.conf. This is where your
; custom incoming call processing should go.
; For sample purposes, this section will read back the
; dialed number and then test DTMF by reading back each
; digit pressed by the caller.
;
; I don't unserstand this part at all. Do I put my phone number here?
; -- MSH
; -
[incoming] ; -- Should match the context you have
; under [voicepulse-in-01] in iax.conf
exten = _NXXNXX,1,Playback(beep)
exten = _NXXNXX,2,SayDigits(${EXTEN})
exten = _NXXNXX,3,Goto(testdtmf|s|1)
;
; This context is used by the sample [arbitrary-name]
; context above to read back each digit you press.
;
[testdtmf]
exten = s,1,Background(beep)
exten = s,2,ResponseTimeout(60)
exten = _x,1,SayDigits(${EXTEN})
exten = _x,2,Goto(testdtmf|s|1)
exten = i,1,Goto(testdtmf|s|1)
exten = t,1,Hangup
; -
; This context is used to send all outgoing calls to the
; VoicePulse Connect! service for connection to the PSTN.
; Asterisk will attempt to dial out through gwiaxt01 first.
; If there is a problem, it will attempt to dial out
; through gwiaxt02.
; YOU MUST HAVE BOTH
[Asterisk-Users] Cisco 7690 Voicemail Problem
Happy New year to you all... I was wondering if anyone can help. I have a couple of 7690's working with the latest SIP image and they call to each other just fine. The problem I have is when I get someones * voicemail. If I have the handset in my hand and am about to leave a message, I get my voice coming out of the 7690 hands free speaker Any Ideas? Thanks Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hmmm - anyone seen this before?
The below is a asterisk message when I try to call from a callerid blocked phone into a SIP (Sipura 3000) FXO gateway - and I have not consciously put any restrictions on incoming calls... Dec 29 10:23:44 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=1a6833c3913bcb6o1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
So we could provide caller position announcements without the callers actually incurring charges? Has anybody tried this (in the UK)? Maybe.. but probably not.. In the UK (and most European countries), the ACCEPT message triggers generation of ringback tone at the calling party's exchange (central office). This is as opposed to the North American way of doing things where the ACCEPT message opens up a one way speech path from the called party to the calling party (originally for providing inband call progress tones I believe). Also, there's a timer on how long you can be in that state without issuing an ANSWER and thus tripping answer supervision/billing commencement. I think technically it IS possible to get UK kit to work in the US fashion, but you have to talk to a switch tech that knows what he's doing, and of course you may get bitten with the Yeah, it's doable, but we don't have that software feature pack installed on our switch line. Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hmmm - anyone seen this before?
Steven P. Donegan wrote: The below is a asterisk message when I try to call from a callerid blocked phone into a SIP (Sipura 3000) FXO gateway - and I have not consciously put any restrictions on incoming calls... Dec 29 10:23:44 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=1a6833c3913bcb6o1 You are using the 3000 trick where you add A to the Caller ID string and then have * strip it off later, (for FXO gateway) aren't you? I assume that you are calling from a cell phone in the 714 area code? I would double check your Sipura PSTN Line settings and make sure that they have a valid login to you * machine. Also, try to do more conventional Sipura FXO call forwarding, not using the A trick. Maybe you can get that to go away. Does the FXO work? I can't imagine that it does, but you never know... -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hmmm - anyone seen this before?
Kristian Kielhofner wrote: Steven P. Donegan wrote: The below is a asterisk message when I try to call from a callerid blocked phone into a SIP (Sipura 3000) FXO gateway - and I have not consciously put any restrictions on incoming calls... Dec 29 10:23:44 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=1a6833c3913bcb6o1 You are using the 3000 trick where you add A to the Caller ID string and then have * strip it off later, (for FXO gateway) aren't you? I assume that you are calling from a cell phone in the 714 area code? I would double check your Sipura PSTN Line settings and make sure that they have a valid login to you * machine. Also, try to do more conventional Sipura FXO call forwarding, not using the A trick. Maybe you can get that to go away. Does the FXO work? I can't imagine that it does, but you never know... -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Well, the FXO does indeed work. And if there is something better to do with the Sipura than the 'A' trick please let me know what it is - I've just been working with the wiki stuff so far. And yes, a cell phone (with callerid blocked) in the 714 area - which unlike the song is having heavy thunderstorms right now... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
So this is doable in the U.S? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick Sent: Wednesday, December 29, 2004 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Music instead of Tunes So we could provide caller position announcements without the callers actually incurring charges? Has anybody tried this (in the UK)? Maybe.. but probably not.. In the UK (and most European countries), the ACCEPT message triggers generation of ringback tone at the calling party's exchange (central office). This is as opposed to the North American way of doing things where the ACCEPT message opens up a one way speech path from the called party to the calling party (originally for providing inband call progress tones I believe). Also, there's a timer on how long you can be in that state without issuing an ANSWER and thus tripping answer supervision/billing commencement. I think technically it IS possible to get UK kit to work in the US fashion, but you have to talk to a switch tech that knows what he's doing, and of course you may get bitten with the Yeah, it's doable, but we don't have that software feature pack installed on our switch line. Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hmmm - anyone seen this before?
Steven P. Donegan wrote: Well, the FXO does indeed work. And if there is something better to do with the Sipura than the 'A' trick please let me know what it is - I've just been working with the wiki stuff so far. And yes, a cell phone (with callerid blocked) in the 714 area - which unlike the song is having heavy thunderstorms right now... Steven, I am sure that I am not the only one who is wondering if the Caller ID is in fact blocked. Does the number that shows up in that error message match your cell number? Without the A trick: http://voxilla.com/forum-viewtopic-t-557.html A trick (voxilla): http://voxilla.com/forum-viewtopic-t-1335.html I have not tried the A trick because it looked like there could be problems such as this. Try the more conventional method. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zapata.conf not being parsed by *
I am running * 1.0.3 for some reason when I start * is does not appear to be parsing my zapata.conf file. I do not see any errors * just does not seem to know to look for zapata.conf. I am unable to use my FXO card to make calls or receive calls. I have been able to configure SIP to work correctly. Any help would be greatly appreciated, I spent most of last night searching for an answer. Thanks Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
Thinking about it we may well be able to do this in the UK as one of the complaints I get about Asterisk is that our ring tone has changed to external callers, (the zone is set correctly for zaptel, but it's different from the normal ring tone), so the tones must be coming from the TE405, not just generated as a result of the accept (unless some data in the accept signifies the tones to generate?) -Original Message- From: Paul Crick [mailto:[EMAIL PROTECTED] Sent: 29 December 2004 18:32 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Music instead of Tunes So we could provide caller position announcements without the callers actually incurring charges? Has anybody tried this (in the UK)? Maybe.. but probably not.. In the UK (and most European countries), the ACCEPT message triggers generation of ringback tone at the calling party's exchange (central office). This is as opposed to the North American way of doing things where the ACCEPT message opens up a one way speech path from the called party to the calling party (originally for providing inband call progress tones I believe). Also, there's a timer on how long you can be in that state without issuing an ANSWER and thus tripping answer supervision/billing commencement. I think technically it IS possible to get UK kit to work in the US fashion, but you have to talk to a switch tech that knows what he's doing, and of course you may get bitten with the Yeah, it's doable, but we don't have that software feature pack installed on our switch line. Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
Thinking about it we may well be able to do this in the UK as one of the complaints I get about Asterisk is that our ring tone has changed to external callers Are you answering the call then doing a Dial? I'd expect if you had exten = 201,1,Dial(SIP/blah) that the caller would just hear regular ringback from their local exchange. We're talking PRI/ISDN right? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RFI: Creating a database of DID providers
Cross posted from asterisk-biz: Is anyone willing to host/manage a website that people can simply browse that lists all current DID providers and their coverage areas? It's a good idea and probably not too hard to implement, it's just a case of deciding how far you want to go.. are areacodes good enough? or do you need to go to NPA-NXX level and start talking about rate centers etc? Ok.. I'm going to have a stab at this.. I'd like to have some kind of search mechanism similar to that at www.voipreview.org where you can select country and area (by state/city? or would people prefer by areacode?) then generate a list of all providers than can supply DIDs in that area, together with setup/rental charges, per minute charges, etc. Before I go reinvent the wheel totally from scratch, is there anyone out there that has data in electronic form that they use already for this sort of thing? I'm looking for country code listings, area code listings, NPA-NXX to city name listings etc. Replies to the list, or forward data files to web-dids at ivrl.com Cheers Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zapata.conf not being parsed by *
On Wed, 29 Dec 2004 13:59:19 -0500, Jerry Rasmussen [EMAIL PROTECTED] wrote: I am running * 1.0.3 for some reason when I start * is does not appear to be parsing my zapata.conf file. I do not see any errors * just does not seem to know to look for zapata.conf. I am unable to use my FXO card to make calls or receive calls. I have been able to configure SIP to work correctly. I've not seen or heard of that before... but the first thing that comes to mind would be some module not being loaded in modules.conf? Since no one has responded yet, thought I'd throw out a shot in the dark :) Leif Madsen. http://www.leifmadsen.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
So this is doable in the U.S? That's what I said wasn't it? ;-) Provided your PRI is set up correctly, you should have a one way speech patch from called party to calling party upon issuance of an ACCEPT. I believe it was done this way to allow PBXs to generate ring back, busy etc, thus offloading the central office switches at either end. This is in contrast to the European way of doing things where call progress indicators are generated at the local exchange (for in-country calls at least). This means that you should be able to generate number not in service or number changed announcements to callers without answer supervision, so that the caller is not charged for the call (which is right, you shouldn't be charged for such announcements). Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] zapata.conf not being parsed by *
Also when I try to dial outbound I get the following errors channel.c:1920 ast_request: No channel type registered for 'Zap' and Unable to create channel of type 'Zap' (cause 66). My assumption is I am getting these errors because Zapata.conf is not being parsed From: [EMAIL PROTECTED] on behalf of Leif Madsen Sent: Wed 12/29/2004 2:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] zapata.conf not being parsed by * On Wed, 29 Dec 2004 13:59:19 -0500, Jerry Rasmussen [EMAIL PROTECTED] wrote: I am running * 1.0.3 for some reason when I start * is does not appear to be parsing my zapata.conf file. I do not see any errors * just does not seem to know to look for zapata.conf. I am unable to use my FXO card to make calls or receive calls. I have been able to configure SIP to work correctly. I've not seen or heard of that before... but the first thing that comes to mind would be some module not being loaded in modules.conf? Since no one has responded yet, thought I'd throw out a shot in the dark :) Leif Madsen. http://www.leifmadsen.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7690 Voicemail Problem
a faint scratching sound of your voice coming out of the speaker? or loud and clear? On Wed, 29 Dec 2004, Paul A Brown wrote: Happy New year to you all... I was wondering if anyone can help. I have a couple of 7690's working with the latest SIP image and they call to each other just fine. The problem I have is when I get someones * voicemail. If I have the handset in my hand and am about to leave a message, I get my voice coming out of the 7690 hands free speaker Any Ideas? Thanks Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Make asterisk launch script after completingcall.
What is contained within asterisk2mp3.py ?
Also, why can't it be like this:
[sip-in]
...
...
...
exten = h,1,System(nice -n 19 asterisk2mp3.py /var/spool/asterisk/monitor
${CALLFILENAME})
instead of calling a Macro? And if you don't record EVERY single
conversation in this context, wouldn't this be executed every single time a
person hangs up, even if nothing was recorded? Maybe a DBPut and an GotoIf
statement?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Polite
Sent: Monday, December 20, 2004 7:46 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Make asterisk launch script after
completingcall.
On sön, dec 19, 2004 at 02:46:00 +0100, Alex Polite wrote:
OK. I now have call recording working for both incoming and outgoing
calls.
Now I want to make those wavs into mp3. I could launch a script from
cron that checks for new wavs and converts them. But that wouldn't be
so elegant.
Launching it from * on hangup would be nicer. How is it done?
Like so:
[sip-in]
exten = 1000,1,SetVar(CALLFILENAME=incoming_${CALLERIDNUM}_${TIMESTAMP})
exten = 1000,2,Monitor(wav,${CALLFILENAME})
exten = 1000,3,Dial(SIP/alex,20)
;exten = 1000,4,Voicemail(u1000)
exten = h,1,Macro(wav2mp3)
[macro-wav2mp3]
exten = s,1,System(nice -n 19 asterisk2mp3.py /var/spool/asterisk/monitor
${CALLFILENAME})
Found it after googling for hours. I have to say that the
documentation for Asterix feels a bit sketchy.
alex
--
Alex Polite
http://polite.se
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Re: [Asterisk-Users] IP Phone recommendations?
On Wed, 29 Dec 2004 10:34:12 -0600, Me wrote: Why not use ATA adapters? This way you can use just about any phone you want. This works well for simple applications, but does not satisfy the business user who needs a multi-line phone with ots of business class features. I love my Polycom IP600s ;-) Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7690 Voicemail Problem
- Original Message - From: Matt Klein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 29, 2004 7:37 PM Subject: Re: [Asterisk-Users] Cisco 7690 Voicemail Problem a faint scratching sound of your voice coming out of the speaker? or loud and clear? I would say a medium crackly version..Actually its the voice from the vmail system ( ' The person at extension blah blah blah') So not too loud but not really clear either Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music instead of Tunes
Paul Crick wrote: So this is doable in the U.S? That's what I said wasn't it? ;-) Provided your PRI is set up correctly, you should have a one way speech patch from called party to calling party upon issuance of an ACCEPT. I believe it was done this way to allow PBXs to generate ring back, busy etc, thus offloading the central office switches at either end. This is in contrast to the European way of doing things where call progress indicators are generated at the local exchange (for in-country calls at least). This means that you should be able to generate number not in service or number changed announcements to callers without answer supervision, so that the caller is not charged for the call (which is right, you shouldn't be charged for such announcements). I do this on our PRIs for some things. You just need to remember that audio is ONE WAY before answer. So you cannot accept anything from the caller. You can only SEND audio to the caller. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] zapata.conf not being parsed by *
You need two files to make the zapata stuff work: /etc/zaptel.conf /etc/asterisk/zapata.conf The first one (/etc/zaptel.conf), configures the Linux driver for the hardware. In theory, you could have another application other than Asterisk using the zaptel driver. The second one (/etc/asterisk/zapata.conf) contains the information Asterisk needs in order to use the zaptel driver. Asterisk uses /etc/asterisk/zapata.conf Linux uses /etc/zaptel.conf Also, did you run modprobe and ztcfg? The zaptel driver won't light up until you give it the spark. [EMAIL PROTECTED] wrote: I am running * 1.0.3 for some reason when I start * is does not appear to be parsing my zapata.conf file. I do not see any errors * just does not seem to know to look for zapata.conf. I am unable to use my FXO card to make calls or receive calls. I have been able to configure SIP to work correctly. Any help would be greatly appreciated, I spent most of last night searching for an answer. Thanks Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/ asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
On Wed, 29 Dec 2004, Paul Crick wrote: So this is doable in the U.S? That's what I said wasn't it? ;-) Provided your PRI is set up correctly, you should have a one way speech patch from called party to calling party upon issuance of an ACCEPT. I believe it was done this way to allow PBXs to generate ring back, busy etc, thus offloading the central office switches at either end. This is in contrast to the European way of doing things where call progress indicators are generated at the local exchange (for in-country calls at least). In Sweden and most countries with EuroISDN the riginating switch will _not_ provide the progress tones to the caller, that is done by the remote switch / end pbx. Of course, this is only done if the pbx at the end signals that in band progress information is available. It is of coursepossible that BT is more prohibitive. They are a bit weird when it comes to isdn. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel
Yes I'm sure that I load wcte11xp. When I do wcte11xp I obtain next error: /lib/modules/2.4.20/misc/wcte11xp.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20/misc/wcte11xp.o: insmod /lib/modules/2.4.20/misc/wcte11xp.o failed /lib/modules/2.4.20/misc/wcte11xp.o: insmod wcte11xp failed Any idea? Srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: miércoles, 29 de diciembre de 2004 18:45 Para: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel On Wed, Dec 29, 2004 at 11:59:58AM +0100, Sergio Serrano wrote: Hi all, I have installed a TE110P in a BOX but when I load zaptel module I can't see any device in /proc/zaptel. And led of the card is green. From /proc/pci, it looks like you pci bus saw the card. Are you sure that you loaded the wcte11xp module? Matthew Fredrickson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
On Wed, 29 Dec 2004, Paul Crick wrote: So we could provide caller position announcements without the callers actually incurring charges? Has anybody tried this (in the UK)? Maybe.. but probably not.. In the UK (and most European countries), the ACCEPT message triggers generation of ringback tone at the calling party's exchange (central office). This is as opposed to the North American way of doing things where the ACCEPT message opens up a one way speech path from the called party to the calling party (originally for providing inband call progress tones I believe). Also, there's a timer on how long you can be in that state without issuing an ANSWER and thus tripping answer supervision/billing commencement. Perhaps UK, but not the European (EuroISDN) way. ACCEPT would be sent when the call is _answered_. PROGRESS / PROCEEDING is sent when the call setup is received by the terinating pbx. If in band progress is indicated most originating switches will open the reverse audio path. This does work in Sweden and I have seen cmments to that effect from Germany as well. Check with your local PSTN provider for their isdn implementatin description. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
On Wed, 29 Dec 2004, Steve Hanselman wrote: So we could provide caller position announcements without the callers actually incurring charges? I doubt it. There is usually a limit on how long a call is allowed to remain in the ALERTING state by the pstn providers. 2-3 minutes are common limits, then the call will be released. Hwever, you can use it for spiffy personalized busy/unavailable messages, for error messages (that number is no lnger available) etc. Works nicely and incurs n charge. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Impossible to compile last version of Asterisk
Now I try to download Asterisk 1.0.3 from ftp.asterisk.org/pub/asterisk and it work again, perhaps the version in cvs has a problem with my configuration... only for information... ciao ciao Paul Paul wrote: Hi, I worked with Asterisk 0.7 without problems until I tryed to load H323. I downloaded the last version and after some try I compile it. I followed the description in /asterisk/channels/h323/Readme and the compilation of this part was good. But the new compilation of Asterisk was impossible (problem with chan_h323.so). I search info with Google and I read that the problem could be with the different kind of version. Today I downloaded the last version of Zaptel, Libpri and Asterisk from cvs and I followed the description in http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation. For Zaptel and Libpri it's ok. but I can' t compile Asterisk... somebody can help me? Thank you very much my version of kernel, I've SUSE 8.1 linux:/usr/src/asterisk # uname -a Linux linux 2.4.21-238-default #1 Thu Jul 29 17:37:30 UTC 2004 i686 unknown linux:/usr/src/asterisk # linux:/usr/src/asterisk # cat /proc/version Linux version 2.4.21-238-default ([EMAIL PROTECTED]) (gcc version 3.2.2) #1 Thu Jul 29 17:37:30 UTC 2004 linux:/usr/src/asterisk # the error make[1]: Leaving directory `/usr/src/asterisk/cdr' make[1]: Entering directory `/usr/src/asterisk/utils' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-HEAD-12/29/04-09:53:32\" -DASTERISK_VERSION_NUM=99 -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -DNO_AST_MM -c -o smsq.o smsq.c smsq.c: In function `main': smsq.c:422: `POPT_ARGFLAG_SHOW_DEFAULT' undeclared (first use in this function) smsq.c:422: (Each undeclared identifier is reported only once smsq.c:422: for each function it appears in.) make[1]: *** [smsq.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/utils' make: *** [subdirs] Error 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fedora Core 3 app_curl compile error?
I ran into the same problem yesterday and installed the libidn-devel package which corrected it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Swan Sent: Tuesday, December 28, 2004 9:29 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Fedora Core 3 app_curl compile error? Hi, I'm making the latest CVS asterisk source on a newly installed Fedora Core 3 distribution. However, when the makefile for asterisk/apps runs, it generates an error when trying to link app_curl.so complaining about not finding -lidn. Has anyone else run into this problem? I can chase down libidn but I find it odd that others on the list have seemingly gotten asterisk to work on FC3 but never complained about this particular problem... Michael Swan Neon Software, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zapata.conf not being parsed by *
Jerry Rasmussen wrote: Also when I try to dial outbound I get the following errors channel.c:1920 ast_request: No channel type registered for 'Zap' and Unable to create channel of type 'Zap' (cause 66). My assumption is I am getting these errors because Zapata.conf is not being parsed Or you have a noload = chan_zap.so in /etc/asterisk/modules.conf or you installed Asterisk before you installed zaptel. Install zaptel before you install Asterisk or the chan_zap modules won't be built. You should also confirm that ztcfg -vvv shows your card and the correct ports. --Eric ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Make asterisk launch script after completingcall.
you can use the g option in the dial to continue even after hangup.
On Wed, 29 Dec 2004 14:46:46 -0500, Paul Rodan [EMAIL PROTECTED] wrote:
What is contained within asterisk2mp3.py ?
Also, why can't it be like this:
[sip-in]
...
...
...
exten = h,1,System(nice -n 19 asterisk2mp3.py /var/spool/asterisk/monitor
${CALLFILENAME})
instead of calling a Macro? And if you don't record EVERY single
conversation in this context, wouldn't this be executed every single time a
person hangs up, even if nothing was recorded? Maybe a DBPut and an GotoIf
statement?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Polite
Sent: Monday, December 20, 2004 7:46 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Make asterisk launch script after
completingcall.
On sön, dec 19, 2004 at 02:46:00 +0100, Alex Polite wrote:
OK. I now have call recording working for both incoming and outgoing
calls.
Now I want to make those wavs into mp3. I could launch a script from
cron that checks for new wavs and converts them. But that wouldn't be
so elegant.
Launching it from * on hangup would be nicer. How is it done?
Like so:
[sip-in]
exten = 1000,1,SetVar(CALLFILENAME=incoming_${CALLERIDNUM}_${TIMESTAMP})
exten = 1000,2,Monitor(wav,${CALLFILENAME})
exten = 1000,3,Dial(SIP/alex,20)
;exten = 1000,4,Voicemail(u1000)
exten = h,1,Macro(wav2mp3)
[macro-wav2mp3]
exten = s,1,System(nice -n 19 asterisk2mp3.py /var/spool/asterisk/monitor
${CALLFILENAME})
Found it after googling for hours. I have to say that the
documentation for Asterix feels a bit sketchy.
alex
--
Alex Polite
http://polite.se
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[Asterisk-Users] Recording/Monitoring a call mid-stream?
Is there a way to monitor a call mid-stream? I did look on the Wiki and found that AstGUI can do it, but its a bit of an overkill. What I want is for a customer service rep, sitting in front of a Cisco 7960, to be able to hit a button (either on their phone, or maybe a specific webpage) that will start recording the call from that point on. Im thinking the services button on the Cisco could be rigged to send the proper command to the manager interface, to start recording the call. But I dont know how to write such a program. Im hoping something already exists. Anybody? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IP Phone recommendations?
[EMAIL PROTECTED] wrote: Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have orphans around. I've got a few different IP phones in my lab (including a C7960), I'm currently loving my Polycom 300 - a solid phone for the price, and everyone says the 500 and 600 are even better. I think I'll be going with Polycom for my customers until the next best thing comes along. I looked at the Snom phones when I was at Astricon, and while they may be technically great, the problem I had with them is that they are not weighted properly. If you've ever yanked your phone off the desk you'll understand the need for a proper ballast. The handsets feel cheap too, because they're too lightweight. Still, from everything I've read you'll certainly want to try one out. Also, the Snom 220 seems to be the best bet as a reception phone, especially if you want a busy lamp field on your swithboard. The Cisco phones are great, but it's hard to stomach paying an extra $100-$300 for that little drawing of the Golden Gate Bridge they put on all their products. One of the exciting things about standards-based telephony is that you can mix and match your phones. It's the same as analog sets; the agony is in the sheer number of choices available. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I was hoping some folks would share what they have found. My primary goal is to replicate the 7310's features and to allow room for growth in the future with telephony applications. One of the big differences between the Norstar and the Asterisk is that the Norstar is a key system, the Asterisk is a PBX. If you completely replace the Norstar your users will will no longer have access to line status on their phones; that is all handled behind the scenes. Also, you will not get busy lamp field, which means you won't be able to monitor who is on the phone (there are ways of doing this in Asterisk, but it's not as intuitive to implement). Finally, the Norstar has hundreds of easy to use features; each one you'll want to keep will need to be carefully hand-crafted in the dial plan. Our primary driver is configurability and features that we can get in Asterisk, that we can get without a lot of money from Nortel. Nortel sure has fallen behind. Even the VoIP stuff they have does not work well, and is barely standards-compliant (if at all). Namely- Voicemail, telecommuting workers on the pbx, better call handling, better automation. I'd like to be able to integrate smart features like directory and call handling to the handset, but I'll freely admit I'm just starting out. My initial goal is to just to get onto Asterisk and get it working. I'll worry about cool stuff later. I think you'll be wise to leave the Nortel KSU in place for a bit. That way you can introduce new features to the users without them also having to learn new phones. There are challenges either way. Our integration and migration plan is as follows: If anyone has some suggestions or pointers I'd love to hear them. 1. Test and evaluate Asterisk with TDM400 with 1 FXO/FSO port each. 2. Configure Asterisk to be the primary PBX and slave the Nortel Meridian system to it using a second TDM400. This avoids immediate replacement of all handsets. Will allow immediate access to features such as Voicemail. 3. Overtime, upgrade desk phones to IP phones. When all phones are replaced, decommission Nortel and sell on Ebay. :) Are you using calling line ID? The problem here is that you have two systems that will each need to wait two rings before answering. The Asterisk will need two rings to get the caller ID, and then it'll take two more to pass the same CLID on to the Norstar. [PSTN]==(2 rings for CLID)==[Asterisk]==(2 rings for CLID)==[Nor*] Make sure you put an autoattendant in the middle, to ensure your callers don't have to wait too many rings before some indication that there's a system at the other end. Also, there is some danger of echo if you put the Asterisk in the middle. You'll want to be patient with this, as it may take a bit of tweaking to sort out. IMPORTANT: Make sure your Asterisk and Nortel are grounded to the same point. Best way to achieve this easily will be to plug them into the same electrical outlet. You do NOT want voltage potentials on the analog loop between the * and Nor*, believe me. The fact is, analog is a technology that really doesn't lend itself well to integration. It can be made to work, but callers and users will have to deal with a lot of extra rings. Also, transfers and the like will involve hookswitch flashes and such. I'm not saying avoid it, just be aware of the need to manage user expectations. One possible way to handle this would be to configure the system so
RE: [Asterisk-Users] zapata.conf not being parsed by *
You know I think the I compiled them in the wrong order. I bet you that is it. I will give it a try and let you know. From: [EMAIL PROTECTED] on behalf of Eric Wieling aka ManxPower Sent: Wed 12/29/2004 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] zapata.conf not being parsed by * Jerry Rasmussen wrote: Also when I try to dial outbound I get the following errors channel.c:1920 ast_request: No channel type registered for 'Zap' and Unable to create channel of type 'Zap' (cause 66). My assumption is I am getting these errors because Zapata.conf is not being parsed Or you have a noload = chan_zap.so in /etc/asterisk/modules.conf or you installed Asterisk before you installed zaptel. Install zaptel before you install Asterisk or the chan_zap modules won't be built. You should also confirm that ztcfg -vvv shows your card and the correct ports. --Eric ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp-0.0.2pre6
Hi Folks, hi Steve I get following error on loading app_rx/txfax.so: ...WARNING[10458]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: symbol errno, version GLIBC_2.0 not defined in file libc.so.6 with link time reference Unable to load app_rxfax.so Spandsp compiled and installed fine The modules for asterisk too. Versions: 1.0.3 for *, libpri, zaptel build from source Distri:debian, testing, kernel 2610 Any hints? THx -- Tho/\/\as ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Recording/Monitoring a call mid-stream?
Hello, You just need to send a simple Monitor command to the Manager interface to start or stop recording(Monitor) on a channel. This can be easily accomplished within PHP or some other basic web scripting language. But you need to have the full channel name to make the recording work. Are you wanting to record Zap, SIP or IAX channels(or all of them)? Will you want the option to record Meetme room conversations? MATT--- -Original Message- From: Paul Rodan [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 29, 2004 3:52 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Recording/Monitoring a call mid-stream? Is there a way to monitor a call mid-stream? I did look on the Wiki and found that AstGUI can do it, but it's a bit of an overkill. What I want is for a customer service rep, sitting in front of a Cisco 7960, to be able to hit a button (either on their phone, or maybe a specific webpage) that will start recording the call from that point on. I'm thinking the services button on the Cisco could be rigged to send the proper command to the manager interface, to start recording the call. But I don't know how to write such a program. I'm hoping something already exists. Anybody? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel
On Wed, Dec 29, 2004 at 09:17:45PM +0100, Sergio Serrano wrote: Yes I'm sure that I load wcte11xp. When I do wcte11xp I obtain next error: /lib/modules/2.4.20/misc/wcte11xp.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20/misc/wcte11xp.o: insmod /lib/modules/2.4.20/misc/wcte11xp.o failed /lib/modules/2.4.20/misc/wcte11xp.o: insmod wcte11xp failed Is that the only digium card you have in that machine? If not, that device I saw on the PCI bus could be another card. What does it say in dmesg about it when you try to load it (as per instructions received upon failure to load module)? Matthew Fredrickson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DSLink modem freeze
Hi Folks, I've been having troubles with a DSL router (DSLink 200E) and SIP phones. When I put any SIP phone (software or hardware) to work behind that DSL router, it completely freeze. I ready tech specs of that DSL router and it says that SIP protocol is supported. ie. I tested two DSLink 200E with the same results. Does anyone has any idea? Thanks in advance. -- Rodrigo P. Telles [EMAIL PROTECTED] Project Manager Devel-IT - http://www.devel.it TDKOM Group ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel
This card is the only card in the system, and other thing, led of the card is fixed green. In dmesg I obtain nothing. Regards, -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: miércoles, 29 de diciembre de 2004 22:31 Para: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel On Wed, Dec 29, 2004 at 09:17:45PM +0100, Sergio Serrano wrote: Yes I'm sure that I load wcte11xp. When I do wcte11xp I obtain next error: /lib/modules/2.4.20/misc/wcte11xp.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20/misc/wcte11xp.o: insmod /lib/modules/2.4.20/misc/wcte11xp.o failed /lib/modules/2.4.20/misc/wcte11xp.o: insmod wcte11xp failed Is that the only digium card you have in that machine? If not, that device I saw on the PCI bus could be another card. What does it say in dmesg about it when you try to load it (as per instructions received upon failure to load module)? Matthew Fredrickson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
