[Asterisk-Users] SIP MWI and MySQL Realtime
I know that there are some patches being worked on to cache realtime users that might ultimately fix this problem, but until then, here is a little script that brings back the MWI when using the excellent mysql realtime architecture with sip: http://www.cheapnet.net/~mike/asterisk/send_mwi.txt This script relies on sipsak utility found at http://sipsak.berlios.de/ Download, rename to send_mwi.pl and chmod 755 it. See top of file for notes on usage and configuration. If you have any feedback, let me know. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Slackware + Asterisk + asterisk-addons
There were some changes recently to the internal structure of the realtime config. Make sure you have the latest CVS copy of both asterisk and asterisk-addons and it should fix the compile error with res_config_mysql. On Sun, 2005-01-30 at 23:16 -0500, Bobby Lacey wrote: > Hello > > I am trying to get asterisk-addons installed so that I can use the > mysql cdr feature. OK, I have the MySQL server (mysqld) installed, but > I noticed that mysql-devel is also required. I tried to compile > asterisk-addons and got a: > > --CUT--- > res_config_mysql.c:422: error: unknown field `realtime_multi_func' > specified in initializer > res_config_mysql.c:422: warning: excess elements in struct initializer > res_config_mysql.c:422: warning: (near initialization for > `mysql_engine') > res_config_mysql.c:423: error: unknown field `update_func' specified > in initializer > res_config_mysql.c:424: warning: excess elements in struct initializer > res_config_mysql.c:424: warning: (near initialization for > `mysql_engine') > res_config_mysql.c: In function `parse_config': > res_config_mysql.c:491: warning: assignment makes pointer from integer > without a cast > /usr/include/asterisk/utils.h: At top level: > res_config_mysql.c:418: error: storage size of `mysql_engine' isn't > known > make: *** [res_config_mysql.o] Error 1 > [EMAIL PROTECTED]:/usr/src/asterisk-addons# > > > So I'm assuming that I need mysql-devel? I went to download the > package from mysql.com, but the only format I saw was .rpm. I am using > Slackware 10.0 so rpm doesnt want to work well. I get the following: > > rpm -Uvh MySQL-devel-4.1.9-0.i386.rpm > warning: MySQL-devel-4.1.9-0.i386.rpm: V3 DSA signature: NOKEY, key ID > 5072e1f5 > error: Failed dependencies: > /bin/sh is needed by MySQL-devel-4.1.9-0 > > > Does anyone have any suggestions or help they could offer? > > Thanks in advance.. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.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] Linux Bridge + QoS Shaper HOWTO available
Asterisk allows setting custom TOS setting, so perhaps you could pick a TOS precedence not normally used, such as CRITIC/ECP or Internetwork Control, in which case your TOS would be 0xB0 or 0xD0 respectively. On Fri, 2005-01-28 at 02:10 +0100, Jasper Spaans wrote: > That could be nice, however, there could be more traffic using tos > 0x10. I wouldn't bet on using it in a config like this. > > On Thu, 27 Jan 2005 15:32:44 -0800, Mike Machado <[EMAIL PROTECTED]> wrote: > > You could also change the tos setting in sip.conf and use a rule similar > > to the one found in the LARTC. This is a bit easier, and perhaps cleaner > > than matching many ports. > > > > # tc filter add dev eth0 parent 1:0 prio 10 u32 \ > > match ip tos 0x10 0xff \ > > flowid 1:4 > > > > That would match TOS of 0x10, which is the "Low Delay" bit enabled. You > > should change this to link to your appropriate flowid and parent node, > > of course. > > > > > > On Thu, 2005-01-27 at 16:39 -0600, [EMAIL PROTECTED] wrote: > > > > > > > > > Does your script work for SIP too? Only for IAX? > > > > > > I'd like to put two of your bridges in my setup – one between the > > > local LAN and the net, and the other at our branch office between the > > > LAN and WAN there. > > > > > > But I need to be able to allow for sip to be QOSed, as we currently > > > have an asterisk server at the remote office, and sip clients here. > > > > > > > > > SIP can easily be added. I am placing IAX2 into high priority by port. > > > You could do this with SIP as well. Add port 5060 and 1-2 to > > > the high priority queue. This HOWTO is meant to be read before > > > implementing it. My example includes shaping video and Citrix, which > > > you may not be interested in doing. In that case you'll need to remove > > > those priority levels. > > > > > > -Ron > > > ___ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.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] Linux Bridge + QoS Shaper HOWTO available
You could also change the tos setting in sip.conf and use a rule similar to the one found in the LARTC. This is a bit easier, and perhaps cleaner than matching many ports. # tc filter add dev eth0 parent 1:0 prio 10 u32 \ match ip tos 0x10 0xff \ flowid 1:4 That would match TOS of 0x10, which is the "Low Delay" bit enabled. You should change this to link to your appropriate flowid and parent node, of course. On Thu, 2005-01-27 at 16:39 -0600, [EMAIL PROTECTED] wrote: > > > Does your script work for SIP too? Only for IAX? > > I’d like to put two of your bridges in my setup – one between the > local LAN and the net, and the other at our branch office between the > LAN and WAN there. > > But I need to be able to allow for sip to be QOSed, as we currently > have an asterisk server at the remote office, and sip clients here. > > > SIP can easily be added. I am placing IAX2 into high priority by port. > You could do this with SIP as well. Add port 5060 and 1-2 to > the high priority queue. This HOWTO is meant to be read before > implementing it. My example includes shaping video and Citrix, which > you may not be interested in doing. In that case you'll need to remove > those priority levels. > > -Ron > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.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] SS7 for *
Have you implemented some kind of device control protocol to a gateway, or do the IMTs terminate on the * box? On Tue, 2004-11-16 at 23:40 +0100, Roger Schreiter wrote: > Hi, > > it is now 3 months ago, that I told here, I were beta testing > SS7 for asterisk. > > I promised to give a result afterwords - here it is: > > There are still some minor problems (maybe more a zaptel > or hardware problem that a SS7 one), but in general it is > running very stable. > > I assume the author will soon present some kind of > licencing model. > > > Roger. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp core dumps asterisk receiving fax
What version of libtiff you have? I had similar problems until I switched down to 3.5.7, and now all faxes work perfectly. Make sure to fully remove older/newer versions before installing a new libtiff. On Mon, 2004-09-20 at 10:02, Maurizio Marini wrote: > debian sid with 2.6.8.1 kernel > at last i was able to get spandsp compiled (see my last post) > now i try to receive a fax but ...core dump!! :( > here the output at console (nothing under /var/log/asterisk): > > Changed from phase 0 to 1 > Slow carrier up > Slow carrier down > Start receiving document > Changed from phase 1 to 4 > Sending ident > >>> CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 > DIS: > Preferred octets: 256 > Can receive fax > Supported data signalling rates: V.27ter and V.29 > R8x7.7lines/mm and/or 200x200pels/25.4mm OK > 2D coding OK > Scan line length: 215mm > Recording length: A4 (297mm) > Minimum scan line time: 0ms > Minimum scan line time for higher resolutions: T15.4 = T7.7 > Get at 9600 > Changed from phase 3 to 5 > Fast carrier up > Coarse carrier frequency 1712.90 (88) > Fast carrier down > Fast carrier up > Coarse carrier frequency 1700.09 (66) > Training error 7.764798 > Training succeeded (constellation mismatch 6.026207) > Fast carrier trained > Fast carrier down > Changed from phase 5 to 4 > Start rx document - compression 2 > Start rx page > >>> CFR: 84 > HDLC underflow in state 5 > Post trainability > Changed from phase 4 to 5 > Fast carrier up > Coarse carrier frequency 1699.83 (66) > Training error 6.394423 > Training succeeded (constellation mismatch 10.554763) > Fast carrier trained > Ouch ... error while writing audio data: : Broken pipe > Segmentation fault (core dumped) > Warning, flexibel rate not > heavily tested! > > some guru can help? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp / compilation errors
Add '/usr/local/lib' to /etc/ld.so.conf if not already, and run 'ldconfig' as root. Then start asterisk. On Mon, 2004-09-20 at 09:45, Graham Turner wrote: > Daniel, thanks for mail back - this has got me much further through spandsp > installation process > > i have progressed through your technote by applying patches to rxfac.c > /txfax.c and applying the Makefile patch > > i assume by rebuild of Asterisk this is make clean; make install in the > /usr/src/asterisk directory ?? - which is as i have done > > do i need to do the same with zaptel / librpi as per the asterisk install > guide ?? > > however i am now in the unfortunate position where the asterisk does not now > start correctly - the console logs the message; > > libspandsp.so - cannot open shared object file - no such file or directory > .. > loading module app_rxfax.so failed ! > > clues on this will be VERY gladly received > > GT > > > - Original Message - > From: "administrator tootai" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[EMAIL PROTECTED]> > Sent: Monday, September 20, 2004 3:54 PM > Subject: Re: [Asterisk-Users] spandsp / compilation errors > > > > Graham Turner a écrit : > > > > >I am attempting installation of spandsp on to an Asterisk installation > on > > >Linux RH9 > > > > > >the distribution i am using is that are URL http://ftp2.tootai.net - the > > >README for which i have followed verbatim - > > > > > > > > It's not a special distribution, it's the original one. It's just here > > as in august the opencall website was down a long time ;-) > > > > >my only issue on this was the target for the port.h / tif_dir.h / > tiffiop.h > > >files in the 'headers' folder of the distribtion > > > > > >i put these in the /usr/include folder based simply on the fact that > there > > >is nothing in the /usr/local/include > > > > > >the tiffio.h / tiffvers.h files are not in here so i am beginning to > suspect > > >the installation of libtiff on the system - however i checked 'rpm -qa' > and > > >it does confirm libtiff 3.5.7 as being installed > > > > > > > > You have libtiff-3.5.7 but what about libtiff-devel-3.5.7 which provide > > tiffio.f and consors? > > > > >any clues on the debug of failed compilation will be gladly received > > > > > >GT > > > > > > > > -- > > Daniel > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Conference Call?
> Another thing you could do is use a regular phone to call into a DID and > enter the conference, then everybody can join that conference and listen. No > bandwidth required, just a phone call to the distributor's Asterisk server. > Then just keep that phone near the person speaking, like a microphone. > I might be able to donate some dial in lines for this. I have local numbers for California, but outside would be a long distance call. I can probably get MP3 streaming of the conference working too. Whoever is organizing the conference can contact me off list if they are interested and don't already have another solution. > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Feature Group D
Yes, this is working for me. The remote side must send *** for this to work. If asterisk does not detect the first *, then it prints the message you are seeing. On Fri, 2004-07-16 at 07:18, Joseph wrote: > Is anyone using Feature Group D with a digium 405P card > and get the ANI spill to work (callerid)? > > I can get the dnis to work fine. > > But not the caller id. > > Here is the error I get: > chan_zap.c:4628 ss_thread: Got a non-Feature Group D input on > channel 3. Assuming E&M Wink instead > > zapata.conf: ** > group = 3 > callerid=asreceived > context=incoming > usecallerid=yes > immediate=no > signalling=featd > channel=1-24 > *** > > zaptel.conf > span=1,1,0,d4,ami > e&m=1-24 > > How can I debug this? > We have tried everything we can think of on the provider side... > Would anyone perchance have a sample of the settings needed > on the provider side to make featd work right? > > Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] accountcode problem
I am trying to set the account code for outbount calls from a sip "friend", but its not working. I have this in my sip.conf: [user1] type=friend host=dynamic secret=test123 context=from-sip-int accountcode=user1 The SIP from: is not "user1" because we allow them to specify their own ANI out our trunks (they are CLEC). Is because the from: not matching the user entry here, it does not which user and therefore does not put the accountcode in the CDR? When I map extensions to this SIP user, the account code is set fine, but I need the account code on outbound calls from this user as well. Thoughts? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: OT: saving/restoring sipura config
> Is the "Sipura Profile Compiler" some perl script that massages the > data? Where can I ftp/http it from? (Or is it some ms-binary, in > which case it wouldn't be all that useful to me.) > It turns the ascii file into a binary config file which the ata can understand. They have binary versions of the spc utility in both windows and unix format. > I'd really prefer just finding a spec describing the config file > syntax/format and the list of variables that I could set. It is > perfectly fine if the downloaded file needs to be binary. It just > needs to be well-enough defined that one can write a BSD or linux > program for. There administration guide documents the options that can go into the config file very well. Check the sipura.com support page. > > -wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Changed IP and subnet now no SIP Register 403
Did you update /etc/hosts? I noticed chan_sip gets upset if it cannot resolve its own hostname. On Sat, 2004-06-12 at 08:40, Steve Totaro wrote: > I built a system and then changed the IP and subnet. Now the phones > will not register, getting a 403. > > Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Did you even read the RFC? Section 10.2.1 clearly talks about adding multiple bindings to the same address-of record. On Sun, 2004-07-11 at 12:31, Paul Mahler wrote: > The whole point of a SIP registration is to identify a UNIQUE device. You > CAN'T HAVE multiple devices registered as the same SIP device. That's WHY > the last device that registers gets the traffic. > > This doesn't have ANYTHING TO DO WITH ASTERISK. This is a SIP issue, not an > Asterisk issue. You should just be happy that Asterisk will do what you > want, even if SIP won't. > > If you really, really want to do this, up the bounty to about $50,000 and > get the SIP specification changed. > > > Paul Mahler > [EMAIL PROTECTED] > Signate, LLC > 665 Third Street > Suite 100 > San Francisco, CA > 94107-1901 > > Asterisk Services and Training > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID and DNIS Problems (Non-Pri T1)
In /etc/asterisk/zapata.conf change signalling=em_w to signalling=featd and restart asterisk On Sat, 2004-07-03 at 17:56, Robert Jackson wrote: > I am trying to receive both CID and DNIS from the telco through a > non-pri T1. Currently I have the T1 setup and operational both outbound > and inbound calls are completed as should be expected. The calls came > in and were placed in the context specified in zapata.conf on exten => > s,1. > > I have requested that the telco provide callerid (they call it ANI) > along with 10 digit dnis for my 800 numbers. I am trying to get > asterisk to recognize these new additions, but for some reason it simply > isn't working. It appears to me that asterisk is taking the digits that > are passed from the telco and using all of them as the DNIS. So when a > call comes in asterisk looks for extension "*CALLERID*800XXX*" where > callerid is the actual number of the calling party. It obviously can't > find a match to that extension and since it changes with each caller I > ended up matching all calls with an exten => _. So I guess my question > is what am I doing wrong? I know that * has to be able to interpret > this information. I am assuming that something is wrong with my > configs. > > Thanks for the assistance, > > Robert Jackson > > zapata.conf > --- > [channels] > > usecallerid=yes > rxwink=300 > echocancel=yes > echocancelwhenbridged=yes > rxgain=0.0 > txgain=0.0 > > ; Local PRI is working fine > switchtype=national > signalling=pri_cpe > context=inbound-local > group=1 > channel => 1-23 > > ; Long distance T1. Receiving *CallerID*DNIS* and interpreting the > ; whole string as the dnis. > switchtype=dms100 > signalling=em_w > context=inbound-longdistance > group = 2 > channel => 25-47 > > zaptel.conf > > loadzone=us > defaultzone=us > > span=1,1,0,b8zs,esf > bchan=1-23 > dchan=24 > > span=2,2,0,b8zs,esf > e&m=25-49 > > extensions.conf > --- > [incoming-local] > include => incoming-main > > [incoming-longdistance] > include => incoming-main > > [incoming-main] > ; Calls incoming to the main administrative line. > exten => 3521234567,1,Answer > exten => 3521234567,2,Queue(administrativeq) > exten => 3521234567,3,Hangup > > exten => 8001234567,1,Answer > exten => 8001234567,2,Queue(administrativeq) > exten => 8001234567,3,Hangup > > ; Calls incoming to the main AR/Patient line. > exten => 1234567890,1,Answer > exten => 1234567890,2,Queue(patientq) > exten => 1234567890,3,Hangup > > exten => 8007654321,1,Answer > exten => 8007654321,2,Queue(patientq) > exten => 8007654321,3,Hangup > > ; Nothing works except this last piece currently. > exten => _.,1,Goto(mainmenu,s,1) > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pager Notification
Another solution if * does not have native TAP support is to use the pager email notification in asterisk, but have the email be sent to your TAP gateway you setup. I have used sendpage (http://sendpage.cpoint.net/) for this purpose before. This of course assumes you want to dial into the page gateway yourself. If you are not worried about it, its probably better to send emails to the public email address provided by most pager/cell phone companies, such as @mobile.att.net. On Thu, 2004-07-01 at 11:03, Chris Travers wrote: > Hi; > > Before I tell a customer that this would require custom development I > figured I would ask here. > > Does Asterisk support pager notification of new voicemails out of the > box? Or do I need an AGI script to do that? > > Also, if I want to call a number from an automated program in Asterisk > and get the DTMF tones entered by the user on the other side, is there > an easy way to do this? > > Best Wishes. > Chris Travers > Metatron Technology Consulting ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SS7 to Pri
> And as far as I can tell, the only way to get callerid etc is > by a PRI to *. I have E&M trunks into * out of our switch. We tell the switch to pass digits in Feature Group D DTMF format, and we are able to get ANI and DNIS. Of course this does not allow to get the calling party's name though. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Users Canreinvite=[yes|no] explained (new subject)
Based on my post yesterday, and the call trace I have, if Asterisk were to make a decision a little differently when sending the the ReINVITEs to phone B in your example (lets say Phone A is the one behind NAT) media might work both directions. In the trace I posted, asterisk first send a reinvite with the private IP of phone A, but then it sent a second reinvite with the visible IP of phone A, which I think would have made the call work in some NAT environments, but then it sent a *3rd* reinvite to phone B, back to the private IP of phone A, breaking the audio from phone B to phone A. What I think happened was when phone A send the 200 OK for its reinvite, Asterisk saw the SDP info from that packet and triggered the 3rd reinvite to phone B, but since nat=1 was on, it should have ignored that SDP, or at least sent the visible IP in the 3rd reinvite and not the private IP. In case you don't have it handy, the call trace I am referring to is here: http://www.cheapnet.net/~mike/asterisk_excel_with_reinvite.log In this log, only the 192.x network is nat'd. 10.x and 172.x have straight routing between them. 10.10.11.77 is the visible IP to 192.168.222.197 according to 10.x and 172.x. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BUG?: reinvite and nat
I have a setup where I am using asterisk a SIP proxy. My ATA is behind NAT. Asterisk user is setup with nat=1 and canreinvite=yes. The call sets up and I can get one way media. The media works ok from the ATA behind the NAT to the external SIP endpoint, but I cannot get media back through the NAT. Calls work just fine if I call from the ATA into an asterisk IVR menu, so I know the NAT router is working properly. Also, if I set canreinvite=no, then things work ok. I see some strange ReINVITEs happening that seem to be the cause of this. When the ReINVITE is sent to the external SIP proxy, it puts SDP parameters out of the raw SDP supplied by the ATA (internal IP), but then issues a second ReINVITE with the correct external NAT IP (detected from the packet source address), but a few moments later it send a 3rd ReINVITE back to the internal NAT IP. What I think is happening is when asterisk sent the ReINVITE toward the ATA, when the ATA issued the 200 response with its SDP, by that time asterisk had sent the 2nd ReINVITE (external IP) to the external SIP endpoint, the 200 reply from the ATA had the internal NAT IP, which was different than it just transmitted to the remote endpoint, so it thought it had to send another 3rd ReINVITE, but this had the internal NAT IP, so it broke media into the NAT from the remote UA. So, I think the bug is that asterisk is sending this 3rd ReINVITE when it should not. I have a trace of all the SIP messages here: http://www.cheapnet.net/~mike/asterisk_excel_with_reinvite.log This is a complicated issue, hopefully I explained it well. In that SIP trace file, the remote SIP UA is 172.20.50.30 (media to .32 and .33), asterisk is 172.20.50.22. The NAT box is 10.10.11.77 on the external interface and 192.168.222.1 on the internal NAT side. The ATA is 192.168.222.197. Between 10.x and 172.x is straight routing (this is internal test network). The only nat is 192.168.222.x is translated to 10.10.11.77 to reach any of the 172.x network. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup problem during intergration with 3rd party pbx
I am trying to integrate asterisk with a 3rd party PBX for voicemail (Mitel). For the most part, things are working well. We only have one main issue left: hangup detection. The connection between the two is: * w/T100P -> Zhone channel Bank FXO port -> Mitel ONS (station) port (Yes, overkill, but we had these parts on hand so we didnt want to buy a TDM FXO card) The channels in * are signaled with fxs_ls signaling, as the PBX can only hand out station lines with the hardware it has. So the problem is when a call from the PBX into * (voicemail) is placed and the user hangs up in the middle of the call, the channel stays in use until voicemail times out. The docs from the PBX state that when a hangup happens, it puts dialtone on the line, which is what is recorded in peoples voicemail boxes when a call is hung up. I have read all the mailing list posts about hangup detection with X100P cards, even got out the volt meter and saw that the line does NOT drop to 0 after a hangup. So the PBX is not acting like a true CO line, but instead puts dialtone on the line. Is there a way for asterisk to detect this dialtone during a call and hangup the channel? This seems to be the only way we are going to solve this hangup problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Prepaid
Have you tried the calling card sample AGI listed in the Wiki? I am not using it in production, but I tested it and it seemed straight forward. http://www.voip-info.org/wiki-Asterisk+tips+and+tricks On Sun, 2004-05-23 at 16:10, usedcanon wrote: > I have a requirement for a setup with prepaid call credits. > > I am aware of the two applications available (been researching for the past > week), app_prepaid and app_rateengine. However neither of the two sound like > exactly what I want. However I was wondering that someone who has used it > might be able to say if they could be used in my scenario. > > Basically my scenario is pretty straight forward. Credit will be allocated > to the ddi, I dont need any announcements etc (maybe low credit warning > during call could be useful thoug). From the users prespective everything > will be transparent. However the call should disconnect when the credit runs > out. The CDR and the account DB need to be adjusted according to the call > made. > > My guess is that app_prepaid could used with modification, I am assuming > here that this is not possible as-is with configuration. > > Basically in case of the prepaid app, the card number can be replace > transparently with the callerID. > > All help, guidence and comments will be extremelly appreciated. > > Umar. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA devices
I have seen several vendors with this type of device. Carrier access and vpacket (now Zhone) are the two that come to mind at the moment. I think I have seen one from Audiocodes as well. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] app_dbmysql and ODBC Voicemail
Oh man, I didn't even notice your vm-odbc.diff before. I hate duplicating efforts. On Fri, 2004-05-14 at 12:58, brian wrote: > I know that, I did a patch for odbc support in app_directory also. > > Bkw > > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Zac Amsler > > Sent: Friday, May 14, 2004 2:11 PM > > To: [EMAIL PROTECTED] > > Subject: RE: [Asterisk-Users] app_dbmysql and ODBC Voicemail > > > > Please be aware that the extension directory does not work when the > > voicemail boxes are not stored in voicemail.conf. > > > > Zac > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Mike Machado > > Sent: Friday, May 14, 2004 1:43 PM > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] app_dbmysql and ODBC Voicemail > > > > > > I have done a little work on asterisk and database integration. Below is > > a link to app_dbmysql, modeled after Brian's app_dbodbc but for pure > > MySQL. > > > > I also ported the mysql-vm-routines.h to ODBC in case anyone is > > interested. > > > > > > You can get both of these from: > > > > http://www.cheapnet.net/~mike/asterisk > > > > > > They were working as of yesterday CVS, but today CVS will not compile > > and I have not looked into why. Let me know if you have any problems or > > feedback with either of them. > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_dbmysql and ODBC Voicemail
I have done a little work on asterisk and database integration. Below is a link to app_dbmysql, modeled after Brian's app_dbodbc but for pure MySQL. I also ported the mysql-vm-routines.h to ODBC in case anyone is interested. You can get both of these from: http://www.cheapnet.net/~mike/asterisk They were working as of yesterday CVS, but today CVS will not compile and I have not looked into why. Let me know if you have any problems or feedback with either of them. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 200
I am using the Snom 105 and am happy with it. I am not doing too many advanced features though. The most advanced feature I use is logging into 2 SIP proxies, and when I make a call, I can choose the outbound line. The only dislike I have about the phone is the handset. It not very comfortable to use and it does not seat well into the phone base. I think the Snom 200 handset looks much better, but thats just from the pictures of it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] alternative FXO gateway to Mediatrix 1204?
On Mon, 2004-05-10 at 12:37, Ernest W. Lessenger wrote: > We use an AudioCodes MP-108 and have been quite happy with it. NOTE: Make > sure you get the most recent software build, the one that came installed on > ours was REALLY old. Might if we ask roughly what you paid for it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stun server
I just put multiple IPs on the same interface and use -a . Seems to work fine. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 DID problem
What signaling are you using in /etc/asterisk/zapata.conf (em, em_w, featd)? When I use a DTMF based signaling, I can see the actual DTMF tones as they are received in my 'full' log. Here is an example of what I see (not real phone number) using a signaling type of 'featd': Apr 30 17:02:46 VERBOSE[47121]: -- Starting simple switch on 'Zap/13-1' Apr 30 17:02:46 DEBUG[47121]: DTMF digit: * on Zap/13-1 Apr 30 17:02:46 DEBUG[47121]: DTMF digit: 5 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 8 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 2 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 6 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 2 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 3 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 2 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 9 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 5 on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: 7 on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: * on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: 7 on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: 0 on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: 4 on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: 5 on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: 9 on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: 7 on Zap/13-1 Apr 30 17:02:49 DEBUG[47121]: DTMF digit: 1 on Zap/13-1 Apr 30 17:02:49 DEBUG[47121]: DTMF digit: 2 on Zap/13-1 Apr 30 17:02:49 DEBUG[47121]: DTMF digit: 0 on Zap/13-1 Apr 30 17:02:49 DEBUG[47121]: DTMF digit: 1 on Zap/13-1 So I can see I am being passed **. This might help you track down if you are actually receiving 4 digit dnis. On Tue, 2004-05-04 at 11:05, Pat Boyle wrote: > Thanks for the reply. > > If I delete the "6" extension and leave the 6020 extension, asterisk > won't answer it and I get the invalid extension message from > asterisk. I suspect that for some reason, the zaptel driver is only > passing forward "6" of the full four digits "6020." > > Any thoughts on why I'm only getting a single digit of the 4 digit > DID? > -Pat > > > On Tue, 2004-05-04 at 11:52, Pat Boyle wrote: > > -- zaptel.conf -- > > span=1,0,0,esf,b8zs > > e&m=1-8 > > loadzone=us > > defaultzone=us > > > > -- extensions.conf -- > > ; Need an extension to pick up calls from the T1 that uses e&m wink > > ; This comes in as a 6 instead of 4 full digits > > ; then pass to the s extension > > exten => 6,1,Wait(1) > > exten => 6,2,Goto(incoming,s,1) > > Get that out of your incoming. You have to match on as many of the > unique digits they are sending to you. Don't include any other > contexts > that might match early. Specifically your incoming should probably > just > contain a list of your DID numbers and a gotos that direct it to the > right sections of the dialplan. > > exten => ,1,goto(Sales-in,s,1) > exten => ,1,goto(Tech-in,s,1) > exten => ,1,goto(vmail,s,1) > exten => ,1,goto(extensions,110,1) > exten => ,1,goto(extensions,111,1) > > Get the picture? With DID you have to be careful not to match too > early, > and this will help you avoid early matches by only being able to match > to the exact DID numbers being sent. > > > > -- zapata.conf -- > > [channels] > > context=incoming > > signalling=em_w > > ; rxwink=600 > > echocancel=yes > > echotraining=yes > > group=1 > > immediate=no > > channel => 1-8 > -- > Steven Critchfield <[EMAIL PROTECTED]> > > - Original Message - > From: Pat Boyle > To: [EMAIL PROTECTED] > Sent: Tuesday, May 04, 2004 9:52 AM > Subject: T1 DID problem > > Hello, > I have a T1 (not PRI) plugged into my Asterisk server with a T100P > card. > > Everything is working well, except I only get the first digit of the 4 > digit DID in Asterisk. The T1 provider (Eschelon) tried switching to > 7 digits, and I only got the first digit of the 7. > > Can anybody help? We're adding another DID and I need to trap it > correctly. > > System info: > Asterisk 0.7.2 > Zaptel 9.1 > Redhat Fedora Core 1 > > Thanks. > > Here are snippets from the relevant files: > > -- zaptel.conf -- > span=1,0,0,esf,b8zs > e&m=1-8 > loadzone=us > defaultzone=us > > -- extensions.conf -- > ; Need an extension to pick up calls from the T1 that uses e&m wink > ; This comes in as a 6 instead of 4 full digits > ; then pass to the s extension > exten => 6,1,Wait(1) > exten => 6,2,Goto(incoming,s,1) > > -- zapata.conf -- > [channels] > context=incoming > signalling=em_w > ; rxwink=600 > echocancel=yes > echotraining=yes > group=1 > immediate=no > channel => 1-8 > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with new sipura firmware 1.0.35a
I have two units running 1.0.35a working just fine. On Mon, 2004-05-03 at 10:08, Victor Perez wrote: > I just tried to upgrade my sipura to firmware 1.0.35a and now I can't connect to it. > It still works but any connection to ports 23 and 80 makes it reboot. Even the flash > tool makes it to crash when trying to connect. Anybody else experiencing this > problem? > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3 encoding of Monitor files
lame did the same thing. The reason I ask this on the asterisk list is that .wav files I record from other sources encode just fine. I think the hitch is the sample rates produced by asterisk. File recorded by gnome sound recorder (lame/bladeenc encode just fine): RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz vs File recorded with Monitor: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz I will give the newer version of sox a try. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3 encoding of Monitor files
I have having problems trying to take a file recorded with Monitor and convert it to MP3. When I use 'play' to play the .wav file, it sounds fine. After bladenc'ing it, it plays at lightening speed, and the voices are all high pitch. I tried using sox to resample to 32000 before encoding, but that didnt work either. Do any of you convert your .wav files to mp3? Monitor call: Monitor(wav|test) 'file' output: test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Sox resample: sox test.wav -r 32000 newtest.wav Bladeenc call: bladeenc newtest.wav newtest.mp3 mpg123 newtest.mp3 # sounds like Im listening in fast-forward mode... Any suggestions on how I can get mp3 versions of files produced by Monitor? On Thu, 2004-04-22 at 15:49, Roscinante wrote: > On Thu, 22 Apr 2004, Dennis Sorge wrote: > > Any recommendations for ripping my .wavs to MP3's? I'm running Mandrake 9.2 > > for a potential music server. Thank you in advance for your suggestions. > > > I use bladeenc, I imagine there is some spiffy front end for it out there > somewhere.. > ___ > Lug-nuts mailing list > [EMAIL PROTECTED] > http://felix.mikesoffice.org/mailman/listinfo/lug-nuts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
> > Not knowing about the switch side of things, but how many interfaces > would a PRI card be able to handle in that switch? I'm betting for $35k > it is quite a few. It may be something to sit down with your copy of the > local tariffs and decide how many circuits over how many months would > pay off that card and make the business pitch to get it. > The card outputs 30 D channels. You still use 24 channels out of your IMT trunks, mapping channel 24 to one of the 30 D channels. Eventually I am sure they will get the card. They were hoping we could make things work in the short term some other way. Might just have to bite the bullet. > Your only other potential solution would probably involve SS7, and that > isn't supported under asterisk now. SS7 on asterisk would be nice for many reasons. Thanks for all your answers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
On Fri, 2004-04-16 at 08:45, mattf wrote: > Sadly it's a limited to PRIs(at least that's what I've been told) You just > can't send that much call data with a good old non-PRI T1. The up side is > you get one extra voice channel to use as compared to a PRI. It's strange > that the carrier doesn't have PRI cabability, I've run into many more > carriers that can't do non-PRI. I've even had one that couldn't send me > ANI(CallerID) on non-PRI lines. > > Once you get into the world of ISDN and PRI, you start to be able to do a > lot more with the signaling of calls, and you even have the ability to do > faster call switching as compared to a non-PRI. > > My advice is to get a different carrier, there are hundreds out there. And > if you absolutely have to have dynamic CallerID transmission you should > verify that your new carrier will let you do that before you sign a > contract(Make sure you verify it by talking to an actual switch tech from > the carrier, sales people will lie through their teeth to get you to sign > that contract). > > Hope this helps, > > MATT--- > My situation is a little different. The carrier switch is about 100 feet away from my asterisk box. The company I am working for is a CLEC and they have their own switch. The switch I am connected to does not have a very expensive PRI signaling card ($35k), so they can only do CAS. If I was a customer and getting service from a carrier, I would definitely have gotten a PRI. Thanks for your perspective on this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
> > > You can usually get CLI on an E&M robbed bit T1 by configuring it right. > Instead of just sending you the DNIS as a string of DTMF they usually > send ***. The DNIS and CLI may be swapped, and there may be > less than 3 *s in the string - wonderful consistency, eh? :-\ I am getting CallerID and DNIS on the inbound calls. What I really need is to be able to set callerID on outbound calls. I am trying to set the callerid using SetCIDNum just before using Dial on a zap channel, but it looks like the switch guys have it set to always stamp the same callerID on the my outbound calls no matter what I put in SetCIDNum or what channel on the T1 I use. Is this a misconfiguration of the switch or a limitation of the signaling protocol? If its the switch, can you give me any pointers as to what I could ask them to look for, or if its the protocol, do you know any other signaling protocol that lets me set outbound callerID (besides PRI)? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
cvs HEAD did infact fix the ringing problem. Thanks Eric! I have another question for all you T1 buffs out there. The T1 I am working with goes into our local phone switch (Excel switch). Currently we are using E & M Wink signaling. The problem is we cannot set callerid on the outbound side. My minimal understanding is that if we had a PRI, I could set the callerID. Unfortunately PRI is one signaling type they cannot do (not have expensive PRI card in switch). So, my question is what other signaling types CAN I set the callerID outbound? My local switch techs cannot seem to answer that question. They just always use E & M for everything. But if I can ask them to specifically try a certain signaling type (such as Feature Group D) or one of the others in the t100p supported list, I could probably get them to change the signaling type on my trunk. Do any signaling types other than PRI support passing outbound callerID? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange T1 Problem
On Thu, 2004-04-15 at 15:26, Steven Critchfield wrote:
> Explicitly answer the line. If that doesn't handle inband audio, there
> is a r flag to dial. This was discussed very recently.
This must be a different problem, because neither of those solutions
worked.
zapata.conf sends call to fixup context:
[fixup]
; Receive call as **
exten => _.,1,Answer
exten => _.,2,Cut(CALLING=EXTEN,*,2)
exten => _.,3,SetCIDNum(${CALLING})
exten => _.,4,Cut(CALLED=EXTEN,*,3)
exten => _.,5,Goto(default|${CALLED}|1)
[default]
exten => 1234567890,1,Answer
exten => 1234567890,2,Dial(SIP/user1|r)
user1's phone rings, but no ring from PSTN caller. user1 picks up, both
can talk ok.
I have been using cvs stable branch. I will try HEAD and see if that
fixes it as suggested by Eric.
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[Asterisk-Users] Strange T1 Problem
When people call into my * box over the T1 interface, they get no ring tone. It rings the SIP phone and when the SIP user picks up, both parties can hear each other ok, its just the PSTN user calling in hears no ring. What could be causing this? I tried setting immediate to yes in zapata.conf, but that causes my DNIS and CallerID to stop being available. T100P with E & M Wink start signaling, all 24 channels are inbound channels (no channel bank or anything like that) to SIP ATAs. The ATA is sending a 180 Ringing reply to the invite, but still no ring. Same symptoms with different vendor ATA devices. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P E&M Wink Trunk
I found ways to do substrings. So this is what I did. I changed the zap
channels to come into a context called 'fixup' and then jump into
default after doing the parsing and setting of CID.
[fixup]
exten => _*NXXNXX*NXXNXX,1,SetCIDNum(${EXTEN:1:10})
exten => _*NXXNXX*NXXNXX,2,Goto(default|${EXTEN:-10:10}|1)
So far things seem to be working well. Does this sound like a good way
to deal with this problem?
On Tue, 2004-04-13 at 17:19, Adam Goryachev wrote:
> > map it in the dial plan. Are there substr functions I can use? Can I
>
> Look at the wiki/tiki www.voip-info.org and search for cmd cut or at
> your console do a show application cut.
>
> If you still don't know what to do, do a search on the wiki for
> variables or read the README.variables in the asterisk source code (docs
> directory).
>
> After all that, if you still can't get it working, tell us what you have
> tried, and someone is sure to adjust your attempts to a working config.
>
> Good luck
>
> Adam
>
>
>
> ___
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[Asterisk-Users] T100P E&M Wink Trunk
I am setting up a box with a T100P. Everything is going well. The
company I am working with has their one phone switch gear. They
provisioned me a E&M Wink T1. Cannot do PRI unfortunately. We chose E&M
so we could pass an unlimited number of DIDs to the trunk as apposed to
FXS loopstart signaling. I can make outbound calls no problem, but I am
having problems with the dial plan for inbound calls. The way they setup
the trunk inbound calls have a dialed number as
"**". I do not know how to parse this out and
map it in the dial plan. Are there substr functions I can use? Can I
just call SetCIDNum on an INBOUND call to get the callerid functions
working?
Here is what I see in the log when a call comes in:
-- Starting simple switch on 'Zap/24-1'
== Unknown extension '*916111*916222' in context 'default'
requested
-- Playing 'ss-noservice' (language 'en')
-- Hungup 'Zap/24-1'
916111 is the calling number (Caller ID)
916222 is the called number
What would be the best way to convert this so I can use just ${EXTEN}?
I can get it to work if I do something like:
exten => _*XX*916222,1,Dial(
but that seems like a hack, plus that does not set callerid. I also dont
know how that work react to callerid being blocked.
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Re: [Asterisk-Users] Spring VON Wrap Up
> Was there any aggressive pricing given for nationwide voip LD? Level3 had several products, one they called Enhanced which was supposed to also include E911 service. They quoted me about $.01 per minute inbound or outbound nation wide. They said they support the top 300 cities in the US and, of course, have plans to serve every rate center in the US. I also went and talked with ITXC, but the rather bad sales person said they were only really interested in international calling and not domestic LD. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Traceroute equivalent
RFC 3261 has previsions for traceroute using the Max-Forwards header. Here is a tool that can take advantage of this feature. http://sipsak.berlios.de/ On Wed, 2004-03-17 at 07:47, David Zuzga wrote: > Is there a traceroute equivalent in the VoIP world? I would like to see the > route a call takes after it gets to the gateway. Basically showing all the > hops until it reaches it's destination or PSTN termination. > > -Dave > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: SIPURA 2000 Problems (Senad Jordanovic)
That is how I upgrade my sipura. I believe I learned that from the manual. I know they update their manual occasionally. Perhaps you have an old version. On Tue, 2004-03-16 at 13:44, Bill Reid wrote: > I have had a similar problem upgrading to .24 . Sipura support suggested > using tftp which worked successfully. > > On the tftp server you use the URL > > http://aaa.bbb.ccc.ddd/upgrade?/path_name/spa.bin > > where aaa.bbb.ccc.ddd is the IP address of the Sipura. > > Do not know why these instructions are not in the manual. > > -- Bill > > > From: "Senad Jordanovic" <[EMAIL PROTECTED]> > > To: "'Stefan Meier'" <[EMAIL PROTECTED]> > > Cc: <[EMAIL PROTECTED]> > > Date: Tue, 16 Mar 2004 16:50:08 - > > Subject: [Asterisk-Users] SIPURA 2000 Problems > > Reply-To: [EMAIL PROTECTED] > > > > > * I can not update device to latest .31 firmware. It just sits > > there waiting for SPA 2000 to "not to be in use". I waited and waited > > for many many minutes... > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Snom 105
I could not find one either, but I did find one for the 200 series phones, which seemed similar. That is where I found the default admin password once I locked myself out. Check the 200 series product page. On Fri, 2004-03-05 at 07:18, Eric Hendrickson wrote: > Greets, > > Apologies for the OT post. I'm working with a Snom 105 and can't seem to > find the "Administrator's Manual" for this phone on Snom's website. Does > anyone know where to find this document? Anyone know how to perform a > "factory reset" on this device? After upgrading the firmware to 2.03o, > it appears that I'm locked out of the administrative menu. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hotel and wireless questions
Would like wireless link contain only voice traffic? If not, it would probably be a good idea to put some sort of minimum bandwidth guarantee and prioritization. I have * running over wireless with such bandwidth management in place and it works fine, but not near the volume I would expect with 3 hotels, so YMMV. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 capability
I believe that lucent tnts idea of ss7 gateway is using IPDC to talk to a box that has the ss7 connection. I know of no open source IPDC implementation. On Mon, 2004-03-01 at 13:38, Michael Baird wrote: > We are looking into turning up SS7, my Lucent TNT's support it by using > a "SS7 Gateway", are there any open source products that will serve this > purpose. I've looked at openss7.org a little bit, it looks kind of > stagnant, and it doesn't appear asterisk has any of the functionality > I'm looking for. I'm hoping someone can give me some suggestions (short > of buying a switch). > > Regards > MIKE > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial via sip gateway?
Bob, I have a question into mediatrix for this, but maybe you have
figured it out. I am trying to map a SIP user to a specific PSTN line. I
have my extensions.conf file as you show below, but on the 1204, it just
grabs whatever line is available, whereas I want extension 101 to always
be port1 on 1204, and extension 102 to be port 2 and so on. I noticed a
NetToPstnSourceFilter MIB per port, and their docs hint at using this,
but the example in the docs describes their FXS to FXO, so I am not sure
what I would put in that MIB. CallerID info? * calling sip extension
number? Have you been able to make this work?
On Sat, 2004-01-31 at 20:22, Bob Knight wrote:
> Rich Adamson wrote:
>
> >I'm having a brain fart
> >
> >What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
> >
> >Been trying stuff similar to:
> > exten => _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
> >where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
> >even try the IP.
> >
> >Rich
> >
> from my extensions.conf:
>
> ;**
> [trunk-local]
> ;**
> exten => _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> exten => _9NXX,2,Congestion
>
> [trunk-toll]
> exten => _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> exten => _91NXXNXX,2,Congestion
--
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Re: [Asterisk-Users] Re: Grandstream Firmware ?
Do both the budgetone and the handytone use the same firmware? On Fri, 2004-01-30 at 06:26, Stephen R. Besch wrote: > Greg Boehnlein wrote: > > > On Thu, 29 Jan 2004, Michael Welter wrote: > > > > > >>I have 1.0.4.45 (beta) on my tftp server. Try it at 66.250.23.58. > >> > >>Cheers, > >>Michael Welter > > > > > > Is there a changelog available for the Beta release train? I'm looking to > > see if they have fixed Early Dial yet. > > > When GS connected to my * server to examine the problem, they promised > to keep me posted on the early dial problem. I haven't heard anything > yet, so I am assuming that it has not been fixed. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream asterisk configuration
On my handytone, if I did not enable STUN, the * box would send the RTP data to my 192.168 address, even though I had nat=yes in sip.conf and the SIP handshake happened with my public IP. It seemed * was not properly translating (ignoring) the IP that was in the SDP into the IP that the SIP handshake happened on. When I enabled STUN, * then started to send the RTP to the correct IP and port, but I still did not have audio. That was as far as I got and have not had time to run additional traces to see what the holdup is now. On Wed, 2004-01-14 at 08:45, SW wrote: > Hi, > > In my experience with GS phones, you need STUN support to make it work > properly (behind NAT), otherwise you would need lot of trial end error to > figure out how to do port forwarding. If you have STUN you wouldn't need to > touch the Netgear (except for firewalls). > > If you can't run your own stun server (need two public IPs) then use one of > many STUN servers out there on public internet. > > For an example enable NAT traversal on your GS phone and point the STUN > server to one of these STUN servers > > larry.gloo.net or stun01.newkinetics.com. > > Then reboot the GS and see how it discover the NAT (top of the gs web GUI). > If it is not a full cone or UDP blocked then you should be fine (Netgear is > restricted cone). > > Cheers > > SW > > > From: "Chandra" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] grandstream asterisk configuration > Date: Wed, 14 Jan 2004 19:35:48 +0545 > Reply-To: [EMAIL PROTECTED] > > i have forwarded ports 5060 and 5000-5008 the ports used by sip and rtp to > grandstream from my netgear. rtp.conf uses rtpstart 5000 and rtpend 5008. i > have also opened all 5060, 5000-5008 ports in my firewall configuration. > grandstream uses 5004 port for rtp. > > what am i missing here? please tell me. > > chandra > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems - FIXED
For the benefit of the archives, here is what I did to fix the problem: Simply reordering allow=ilbc allow=ulaw to allow=ulaw allow=ilbc in sip.conf Fixed the problem. I guess the Handytone is not that graceful in codec negotiation? I did notice * sends RTP packets with payload type 97, which ethereal says is an unknown type. * also sends the payload 97 to sipura, but after a few packets, it starts to send ones with payload type 97, which ethereal identifies as "ITU-T G.711 PCMU", maybe after it notices sipura is sending them payload type 0 packets. In the case of handytone, handytone sends NO RTP packets to *, so maybe it does not get the clue. On Sun, 2004-01-04 at 20:16, Mike Machado wrote: > I am trying to get the handytone 286 to make a very simple call to * and > having problems. It registers with * just fine, but when I place a call > (to echo test, for example), the RTP stream seems to have problems > opening. Here is there error I get in *: > > WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries > exceeded on call [EMAIL PROTECTED] for > seqno 0 (Response) > > When doing traces with ethereal, I see successful SIP and SDP > handshakes, but when * sends handytone RTP packets, I see a ICMP Port > Unreachable messages sent from Handytone to * regarding the UDP RTP > packet. * then gives up and I see a BYE from *, which handytone acks. > > Handytone config is default except obvious SIP registration parameters. > I also have a Sipura SPA2000 and everything works perfect for that one, > same extension and everything (not at same time of course). > > sip.conf entry: > > disallow=all; Disallow all codecs > allow=ilbc > allow=ulaw ; Allow codecs in order of preference > > [131] > type=friend > host=dynamic > reinvite=no > canreinvite=no > qualify=300 > callerid="handytone <131>" > mailbox=131 > nat=0 > > > Handytone info: > > Software Version:Program--1.0.4.17Bootloader--1.0.0.11 > HTML--1.0.0.19 > > > Both on same subnet, no NAT. I have two Handytones, both exhibit same > symptoms. > > Anyone else have this problem? -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems
I guess that was another thing that was strange. When I talked, I saw no RTP coming from the handytone to *. Would there be a reason the handytone would not send RTP until it successfully received a RTP packet from *, but since its not accepting RTP, it would not send it either? I do not even get one way communication, I get no way communication. Does anyone out there have this firmware version of Handytone working at all with *? On Sun, 2004-01-04 at 20:55, John Baker wrote: > I had a similar problem with a Cisco phone, i.e., the "Maximum retries > exceeded on call" error. > It took three days to track down the error to buggy network hardware. > > Same symptoms, too - phone registered, one way conversation was ok (had a > test extension > for music on hold) > > Fixed the hardware, phone works great. > > John > > ----- Original Message - > From: "Mike Machado" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Sunday, January 04, 2004 10:16 PM > Subject: [Asterisk-Users] Grandstream Handytone 286 RTP Problems > > > > I am trying to get the handytone 286 to make a very simple call to * and > > having problems. It registers with * just fine, but when I place a call > > (to echo test, for example), the RTP stream seems to have problems > > opening. Here is there error I get in *: > > > > WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries > > exceeded on call [EMAIL PROTECTED] for > > seqno 0 (Response) > > > > When doing traces with ethereal, I see successful SIP and SDP > > handshakes, but when * sends handytone RTP packets, I see a ICMP Port > > Unreachable messages sent from Handytone to * regarding the UDP RTP > > packet. * then gives up and I see a BYE from *, which handytone acks. > > > > Handytone config is default except obvious SIP registration parameters. > > I also have a Sipura SPA2000 and everything works perfect for that one, > > same extension and everything (not at same time of course). > > > > sip.conf entry: > > > > disallow=all; Disallow all codecs > > allow=ilbc > > allow=ulaw ; Allow codecs in order of preference > > > > [131] > > type=friend > > host=dynamic > > reinvite=no > > canreinvite=no > > qualify=300 > > callerid="handytone <131>" > > mailbox=131 > > nat=0 > > > > > > Handytone info: > > > > Software Version:Program--1.0.4.17Bootloader--1.0.0.11 > > HTML--1.0.0.19 > > > > > > Both on same subnet, no NAT. I have two Handytones, both exhibit same > > symptoms. > > > > Anyone else have this problem? > > > > > > -- > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream Handytone 286 RTP Problems
I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 0 (Response) When doing traces with ethereal, I see successful SIP and SDP handshakes, but when * sends handytone RTP packets, I see a ICMP Port Unreachable messages sent from Handytone to * regarding the UDP RTP packet. * then gives up and I see a BYE from *, which handytone acks. Handytone config is default except obvious SIP registration parameters. I also have a Sipura SPA2000 and everything works perfect for that one, same extension and everything (not at same time of course). sip.conf entry: disallow=all; Disallow all codecs allow=ilbc allow=ulaw ; Allow codecs in order of preference [131] type=friend host=dynamic reinvite=no canreinvite=no qualify=300 callerid="handytone <131>" mailbox=131 nat=0 Handytone info: Software Version:Program--1.0.4.17Bootloader--1.0.0.11 HTML--1.0.0.19 Both on same subnet, no NAT. I have two Handytones, both exhibit same symptoms. Anyone else have this problem? -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Port density: DS3 cards?
My understanding is that the ascend gear only speaks IPDC and not MGCP, so not sure it would even work with asterisk. On Thu, 2003-12-04 at 15:09, Steve Dolloff wrote: > I would be seriously wary of putting a DS3's worth of voice traffic on a > TNT. I don't believe they are rated to handle that much voice. The > APX1000 would be a much better platform, but I don't know if you can > find one used. > > Stephen > > > -Original Message- > > From: Ernest W. Lessenger [mailto:[EMAIL PROTECTED] > > Sent: Thursday, December 04, 2003 4:51 PM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] Port density: DS3 cards? > > > > At 02:34 PM 12/4/2003, you wrote: > > >However, considering the traffic volumes that you are talking about, > is > > it > > >really true to say that the traditional telco cards are > astronomically > > >priced, given the amount of revenue that can be generated per month > on a > > >DS3? > > > > Eight quad-span T-1 cards from Digium: $8,970 > > Three reasonable-quality asterisk servers: $1,000 > > One T-1/DS-3 MUX: $5000 > > > > Total system cost: $14,970 > > > > That actually sounds quite reasonable to me. However, if I were doing > this > > myself I would look hard at getting a MAX TNT with VoIP capability off > > eBay. The price would be equivalent or less, the interface would be > more > > complicated, but all the DSP would be done by the MAX. > > > > --Ernest > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
