Re: [asterisk-users] Sending SMS
Kannel + AGI seems like the best solution as long as you have a SMPP connection. Has anybody been able to send MMS thru Kannel? (Integrated with Asterisk, of course). I know there's MBUNI for Kannel but is there a Gateway out there for people with no SMPP connectivity? Jesus Mogollon On Tue, Mar 11, 2008 at 10:14 PM, troxlinux [EMAIL PROTECTED] wrote: where I can find information?, I continue looking for in google and I don't find a lot greetings 2008/3/11, monim benayad [EMAIL PROTECTED]: We use : Kannel + AGI ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chanspy severe sound problems
I'm also experiencing the same problem. Has anyone found a fix for this? Jesus On 2/7/07, Santiago Aguiar [EMAIL PROTECTED] wrote: Hi everyone! I'm using Asterisk 1.2.7.1 on a CentOS 4 server with 5 - 9 agents and I'm having some issues with the Chanspy application. All the agents are on SIP channels with g711 and all the communications are inside a LAN. When I'm spying a SIP channel, the audio from one of the ends (normally the caller) sounds *extremely* (unusable) choppy, as if it was losing some frames. Sometimes the called party is heard almost perfectly, but there are ALWAYS sound quality issues. The agents do not report any problem, and the audio recorded with the Monitor applications sounds reasonably fine. I'm able to reproduce the problem with any amount of load and it happened also while doing tests with my computer as an Asterisk server. Additional Information: * Asterisk 1.2.7.1 built by test @ ast3 on a i686 running Linux on 2006-04-24 10:52:49 UTC * Linux foo.bar.com 2.6.9-34.0.2.ELsmp #1 SMP Fri Jul 7 19:52:49 CDT 2006 i686 i686 i386 GNU/Linux * Intel(R) Pentium(R) 4 CPU 3.00GHz, 1GB RAM. did anyone encountered the same situation? Google only reported one similar problem without a solution ( http://bugs.digium.com/print_bug_page.php?bug_id=7340) any ideas are welcome! thanks a lot! saludos, -- santiago aguiar *netlabs* * Palmar 2548 Montevideo, Uruguay Tel. +(598 2) 707 7687 Fax. +(598 2) 709 4866 * http://www.netlabs.com.uy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing a variable from one Asterisk box toanother
Hi Craig
I've been developing a Recording Server app (which I will be giving back
to the community) and one of the requirements is for the recording to be
offloaded to several machines. Because of the filename is being set prior to
the recording, I need to pass this variable to the slave server. I'm using
1.2.13 (heavily patched) and I came across your email. Any chance of getting
your port? Thanks for your help...
Jesus Mogollon
On 2/22/07, Craig Guy [EMAIL PROTECTED] wrote:
Hi Richard,
there was a thread regarding this a while ago on the dev list which
resulted
in a patch being made to allow variable passing via IAX2 channels. See
http://bugs.digium.com/view.php?id=7619 for the patch which I think is in
SVN or anyhow, is not in 1.2
I have recently backported this patch to 1.2 and have a patch which is
tested against 1.2.12, 1.2.12.1 and 1.2.15, but should work against at
least
1.2.13 and 1.2.14. The patch introduces a new dialplan function called
IAXVAR, Email me if interested.
Craig
- Original Message -
From: Richard Lyman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 21, 2007 7:27 AM
Subject: Re: [asterisk-users] Passing a variable from one Asterisk box
toanother
Richard Lyman wrote:
Eric Bishop wrote:
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All
works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call dials into Box 2 the variable Foo does not get passed...
Does anyone have any clever ideas?
as noted in asterisk/docs/README.variables (iirc)
you should see that variable inheritance can occur by prefacing the
variable with '_' or '__'
also, depending on the age of your asterisk you might want to start
using
'Set' vice 'SetVar'
also, having ${EXTEN:0} , the :0 doesn't do anything, so you should not
use it and just have ${EXTEN}
i hope this helps
sadly replying to my own post, but, i forgot to mention that
passing variables with IAX2 can be an issue sometimes when you use
user and peer (the user side can pass vars the peer side can not, or
doesn't accept them iirc)
this does not happen using friend, but that has its own issues... check
the wiki for more thoughts about this.
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Re: [asterisk-users] Asterisk Java w/ Threads
Let me rephrase that: It would be a lot *simpler*. From the programming point of view you are connecting to a single events source. Astmanproxy is very stable as well. On 3/3/07, Stefan Reuter [EMAIL PROTECTED] wrote: Jesus Mogollon wrote: The best option would be to use AstManProxy and connect your event manager to it. why would adding a new system in between be better than directly connecting to multiple Asterisk servers? =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] Steuernummern 215/5140/1791 USt-IdNr. DE220701760 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Java w/ Threads
The best option would be to use AstManProxy and connect your event manager to it. Jesus On 3/2/07, Doug Garstang [EMAIL PROTECTED] wrote: Ok, so I ain't much of a Java programmer, but... Can the Asterisk Java API be written with threads? Ie, I need to connect to multiple Asterisk systems from the one java application. I tried to make my class which implements ManagerEventListener, also implement Runnable, but got errors because the Runnable interface doesn't throw InterruptedException. Anywho... Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Motherboard 3.3V PCI for TE412P
Hi all Does anyone know of any motherboards with PCI slots that can take the TE412P card? Is there such a MB for Athlon 64 or P4 procs? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unicall reload problem
I'm have a problem with Unicall not being able to recover from an Asterisk reload. When I try reloading, Unicall reports:Jul 24 22:50:44 ERROR[9252]: chan_unicall.c:3444 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 0, channel = 1Jul 24 22:50:44 ERROR[9252]: chan_unicall.c:4216 setup_unicall: Unable to register channel '1-15'Jul 24 22:50:44 WARNING[9252]: chan_unicall.c:4536 reload: Reload of chan_unicall.so is unsuccessful! Why would this be the case?Jesus Mogollon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple periodic announcements in queues? Possible?
This patvh allows you to do just thathttp://bugs.digium.com/view.php?id=6681On 5/5/06, A.J. Paxson [EMAIL PROTECTED] wrote:Hi all!Curious if there was a way to introduce multiple announcments in the queues.Rather than just:Thank you for holding.Your call is important to us...wait 60 sec...[repeat]Is it possible to:Thank you for holding.Your call is important. ...wait 60 sec...Did you knowyou can.wait 60 sec...We are currently experiencing high call volumeswait 60 sec...[repeat]Maybe a:Periodic-announce = AGI(rotate-msg) Periodic-announce-frequency = 60Thanks for any advice!!~~Aaron___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error installing asterisk
yum install libidn-develOn 4/19/06, Luis Fernando Ramírez Cueva [EMAIL PROTECTED] wrote: I am instaling asterisk on Fedora core 3. I have instaled zaptel-1.2.3, libpri-1.2.2, but when I am instaling (make install) asterisk I have the following error:_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o app_zapscan.o app_zapscan.c gcc -shared -Xlinker -x -o app_zapscan.so app_zapscan.ogcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o app_page.o app_page.c gcc -shared -Xlinker -x -o app_page.so app_page.ogcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o app_curl.o app_curl.cgcc -shared -Xlinker -x -o app_curl.so app_curl.o -L/usr/lib -lcurl -L/usr/kerberos/lib -lssl -lcrypto -lgssapi_krb5 -lkrb5 -lcom_err -lk5crypto -lresolv -ldl -lz -lgssapi_krb5 -lkrb5 -lk5crypto -lcom_err -lresolv -L/usr/kerberos/lib -lidn -lssl -lcrypto -lssl -lcrypto -lgssapi_krb5 -lkrb5 -lcom_err -lk5crypto -lresolv -ldl -lz -lz /usr/bin/ld: no se puede encontrar -lidncollect2: ld devolvió el estado de salida 1make[1]: *** [app_curl.so] Error 1make[1]: Leaving directory `/usr/local/src/asterisk-1.2.4/apps'make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk-1.2.4]#I am not sure what happend. Someone can help me please.I have:**openssl-0.9.7a-40xmlsec1-openssl-1.2.6-3openssl-devel-0.9.7a-40pyOpenSSL-0.6-1.p23**ncurses-devel-5.4-13ncurses-5.4-13***/*** zlib-devel-1.2.1.2-1zlib-1.2.1.2-1bison-1.875c-2 LLama Gratis a cualquier PC del Mundo.Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail - direct call
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMailIt's all in thereOn 2/13/06, Tomislav Parčina [EMAIL PROTECTED] wrote: Hi list!How to send a call directly to voicemail recording?When I put thisexten = 313,n,VoiceMail,u221Or thisexten = 313,n,VoiceMail,b221In my dial plan, calling person first hears greeting message (busy or unviable). I would like to avoid greeting message (I would play something with Playback application). Is it possible? --Tomislav Parcinatparcina#lama.hr___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dundi key Problem
Don't use the file extension (get rid of .pub when defining the key in dundi.conf). It will work.Jesus MogollonGlobal IP Systems, LLChttp://www.globalipsystems.com On 2/1/06, Jonathan k. Creasy [EMAIL PROTECTED] wrote: I am getting the following message when trying to lookup up a number viaDundi:Feb1 13:39:24 NOTICE[20146]: pbx_dundi.c:1309 update_key: No such key'office.pbx.bluegrass.net.pub' for creating RSA encrypted shared key for '00:a0:c9:55:91:89'!I have created keys on each box with astgenkey -noffice.pbx.bluegrass.net using the host name for each box of course.I then copied the .pub files to the /var/lib/asterisk/keys folder from each box to the other box.What am I missing?-Jonathan___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FAX Problems - PRI, Adtran and ZetaFax
Greetings I've been banging my head against a wall for over a week about a problem with faxing and asterisk. I have ZetaFax (a faxing application for Windows) connected via modem to a channel bank. The fax request (incoming) comes in thru the PRI, gets sent to the FXS port where the fax system is connected and Asterisk picks up. No prob there. I call the number and I hear the faxing tones. My problem is that, if I get a multi-page request, pages get cut off (most of the time). Sometimes they get cut off during the middle of the transmission, some earlier. None get past (I'd say) 70% of the full document being transmitted. in my zapata.conf I have (for the FXS fax port): signalling=fxo_ks context=app-fax-out-test adsi=no callerid=3052545900 callwaiting=no echocancel=no echocancelwhenbridged=no faxdetect=both group=10 rxgain=10.5 txgain=-5.0 accountcode=avi channel=73 I have two 4-port T1 cards in this machine (dual xeon, 1 gig of RAM). Voice calls are perfect. I was reported that, under the old system, everything worked perfectly (also an asterisk system, albeit a very old one). Has anyone hit a wall like this? Any help will be *REALLY* appreciated! Jesus Mogollon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] R2-Digital (Q.421)
That's exactly how they explained it works. The DTMF is only to provide DNIS and using the register signalling. When I run testcall, I get the handshake but when testcall sends the first digit, the remote equipment doesn't recognize it (because it's expecting a DTMF signal) and then it times out. I don't want to buy the Cisco for this :( especially when I wanted it to be a full open source solution. What would it take for you to implement it? 2005/11/10, Julio Arruda [EMAIL PROTECTED]: Just to clarify this in my head :-)..So...They are using E1/R2 (the R2 Digital)in fact, for all the line signaling(nothing unusual)The register signaling, that I was under impression would be MF in each timeslot (MFC5C in .br, not sure if the same in others), is in fact DTMFin this trunk, and only to provide DNIS ?(in Brazil R2, the register signaling has some collect call informationand etc).Steve Underwood wrote: Hi, I tried hunting for a little more info. I think all that happens with this is they use the Q.421 spec for handling the ABCD bits, and then simply send the DNIS through as DTMF after the seize if acknowledged. That means they loose some of the functionality of real R2 signalling - e.g. no busy, NU, or congestion detection. It wouldn't take a lot of work to implement that. Regards, Steve Steve Underwood wrote: Hi Jesus, The Cisco kit, and one or two other products, offer an R2 digital using DTMF mode, but this is the first time I have heard of it being used. The spec for this is definitely not Q.421. That spec does not mention DTMF at all. R2 using DTMF doesn't appear to be in the ITU specs, as far as I can tell. Without a spec, or any equipment to play with, there isn't a lot I can do right now. Steve Jesus Mogollon wrote: Hi Steve:Thanks for your help. I really appreciate it.. My provider is CANTV in Venezuela. There's a venezuelan variant in the code and I'm using that. Incoming works perfectly, outgoing is not working. I'm being told that incoming is MFCR2 but outgoing is R2-Digital with DNIS DTMF. There is a Cisco router working and it's using the following: r2-digital-dtmf-dnis R2 ITU Q421 DTMF tone signaling with DNIS What's the equivalent in libmfcr2 and Unicall? Again, thank you for your help and your code! Jesus Mogollon 2005/11/5, Steve Underwood [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi Jesus, FX is not a variant of R2. It is a completely different signalling protocol. This means your service provider is using R2 for some of your channels, and providing all your incoming calls on those channels. It is use FX signalling for other channels, and you must make your outgoing calls there. Someone else told be about a similar configuration. I think they were able to use chan_zap for the other channels, and make use of its FX signalling features. I am not sure how that works, as FX signalling over E1s is far from standardised. Regards, Steve Jesus Mogollon wrote: Steve: That's exactly what I'm using. Incoming calls work like a charm but when I try calling I get a protocol error. My provider says that for outgoing I need to use fx signalling. I see that in unicall.conf there's such a thing as protocolvariant=fx but if I uncomment that line, unicall gives me an error. Any ideas? Thanks for your help... 2005/11/4, Steve Underwood [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Jesus Mogollon wrote: Does anyone know how to make this work with Asterisk? (R2-Digital (Q.421)) I have MFCR2 configured but I'm told that outgoing calls are to use Q421 R2 Digital signalling. Any help is appreciated.Jesus Mogollon See http://www.soft-switch.org http://www.soft-switch.org Steve ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.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] MFC/R2
in unicall.conf set the amount of channels you need in a group and then call
Dial(Unicall/g1/${EXTEN})2005/11/10, Bruno de Assumpção Loureiro [EMAIL PROTECTED]:
Hi users,how can I do a group of unicall channels to round? Is there somethinglike Dial(Unicall/R1/...) ??Sometimes a outbound channel is locking up. It isn't happening withthe incoming calls.
I'm using unicall-pre0.0.5.Best Regards,Loureiro.--Bruno de Assumpção Loureiromsn: [EMAIL PROTECTED]___
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Re: [Asterisk-Users] R2-Digital (Q.421)
Hi Steve: Thanks for your help. I really appreciate it.. My provider is CANTV in Venezuela. There's a venezuelan variant in the code and I'm using that. Incoming works perfectly, outgoing is not working. I'm being told that incoming is MFCR2 but outgoing is R2-Digital with DNIS DTMF. There is a Cisco router working and it's using the following: r2-digital-dtmf-dnis R2 ITU Q421 DTMF tone signaling with DNIS What's the equivalent in libmfcr2 and Unicall? Again, thank you for your help and your code! Jesus Mogollon2005/11/5, Steve Underwood [EMAIL PROTECTED]: Hi Jesus,FX is not a variant of R2. It is a completely different signallingprotocol. This means your service provider is using R2 for some of yourchannels, and providing all your incoming calls on those channels. It is use FX signalling for other channels, and you must make your outgoingcalls there. Someone else told be about a similar configuration. I thinkthey were able to use chan_zap for the other channels, and make use of its FX signalling features. I am not sure how that works, as FXsignalling over E1s is far from standardised.Regards,SteveJesus Mogollon wrote: Steve: That's exactly what I'm using. Incoming calls work like a charm but when I try calling I get a protocol error. My provider says that for outgoing I need to use fx signalling. I see that in unicall.conf there's such a thing as protocolvariant=fx but if I uncomment that line, unicall gives me an error. Any ideas? Thanks for your help... 2005/11/4, Steve Underwood [EMAIL PROTECTED] mailto: [EMAIL PROTECTED]: Jesus Mogollon wrote: Does anyone know how to make this work with Asterisk? (R2-Digital (Q.421)) I have MFCR2 configured but I'm told that outgoing calls are to use Q421 R2 Digital signalling. Any help is appreciated. Jesus Mogollon See http://www.soft-switch.org Steve___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] libmfcr2 - spandsp.h: present but cannot be compiled
Hi all When I try compiling libmfcr2 I get: spandsp.h: present but cannot be compiled Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] R2-Digital (Q.421)
Does anyone know how to make this work with Asterisk? (R2-Digital (Q.421)) I have MFCR2 configured but I'm told that outgoing calls are to use Q421 R2 Digital signalling. Any help is appreciated. Jesus Mogollon ___ --Bandwidth and Colocation sponsored by Easynews.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] R2-Digital (Q.421)
Steve: That's exactly what I'm using. Incoming calls work like a charm but when I try calling I get a protocol error. My provider says that for outgoing I need to use fx signalling. I see that in unicall.conf there's such a thing as protocolvariant=fx but if I uncomment that line, unicall gives me an error. Any ideas? Thanks for your help...2005/11/4, Steve Underwood [EMAIL PROTECTED]: Jesus Mogollon wrote:Does anyone know how to make this work with Asterisk? (R2-Digital(Q.421)) I have MFCR2 configured but I'm told that outgoing calls areto use Q421 R2 Digital signalling. Any help is appreciated. Jesus MogollonSee http://www.soft-switch.orgSteve___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.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] User language switching in dial plan
Well, that's why you'd see languages refered to as ES_mx ES_es ES_pe ES_ve And so on much better way to deal with language variants... 2005/11/4, Andres Tello Abrego [EMAIL PROTECTED]: Mexican Spanish..Ha, funny term...:)Mexican Spanish = mx from MeXico...es = ESpain...So, es would be.. humm. Espain Spanish?Chuck Bunn wrote: Hi, What is the best way to allow a user to select the language they hear in the dial plan?In other words I want the phone to answer Hello welcome to ABC company to continue in English press 1 Followed by the same thing in Spanish (Mexican Spanish - I live in the South West United States) but with a press 2. What I would like to avoid is creating two different dial plans and it looks like I can do this, does the following look correct?? By the way is there any prerecorded language selector similar to the above? Or at least something like 'to continue in English press' and 'to continue in Spanish press' the later being in Mexican Spanish. Also I could not find a designator for Mexican Spanish is 'es' correct?? [language] exten = s,1,Answer() exten = s,2,Background(enter-language-extension) exten = 1,1,Set(LANGUAGE()=en) include = internal exten = 2,1,Set(LANGUAGE()=es) include = internal [internal] exten = s,1,Background(enter-ext-of-person) exten = 101,1,Dial(zap/1,10) ... Thanks ___ --Bandwidth and Colocation sponsored by Easynews.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___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unicall
Hi has anyone used MFCR2 using Unicall? I need to use the protocol_variant=fx but Asterisk crashes saying that there isn'ty such a module, though it appears as an option in the configuration file. Does anyone know why it isn't working? Jesus Mogollon ___ --Bandwidth and Colocation sponsored by Easynews.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] spandsp patch
http://zarzamora.com.mx/asterisk/48 I know it´s a tutorial for E1/R2 but it has a new patch that works with CVH-HEAD and spandsp. Try it out.2005/10/31, Andy Kuo [EMAIL PROTECTED] :Hi all, I'm trying to install spandsp. I followed the instructions on http://www.soft-switch.org/installing-spandsp.html , and when I applied the patch, I got the following errors: [EMAIL PROTECTED] apps]# patch apps_makefile.patchpatching file MakefileHunk #1 FAILED at 55.Hunk #2 FAILED at 93.2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej I tried spandsp-0.0.2pre19 on Asterisk 1.1x and also on Asterisk 1.0.9. I also tried spandsp-0.0.2pre21a on Asterisk 1.1x, but nothing worked for me. I'm not a C programmer, and I really don't know what to modify in the patch. I searched around, but didn't really find anything on it. Can anyone please help? Thanks. AK ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Please recommend a phone
Hi All: I'm in need of a phone that would blink a led to let the callee know that there is an incoming call. The GXP-2000 does this but I want an alternative to Grandstream. Any help is appreciated. ___ --Bandwidth and Colocation sponsored by Easynews.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] Grandstream GXP-2000 Poor sound Quality
Greetings We have all those problems and then some... after a while, the phone starts degrading: The ringing becomes lower and lower and there is a lot of stuttering in the conversation. Also, if I stop/start asterisk, half of the phones reconnect while the rest don't. I was using the same firmare as you but had to roll back to 1.0.1.9 because of the degrading issue. We have some polycoms connecting to the same server and they have no problems whatsoever so we know it's a problem with the GXP. These phones are definately NOT ready for prime time. I would stay away from them. Play it safe and use Polycoms or, if too expensive, maybe Sipuras 841. These GXP-2000s are pure evil. Jesus Mogollon Global IP Systems, LLC http://www.globalipsystems.com2005/9/1, Aaron W [EMAIL PROTECTED]: I have upgraded the GXP-2000 to the newest firmware 1.0.1.12 and the phone is much more usable However, I still have two slight sound quality issues: 1) There is static on the line at all times. It is not that noticable to me, but when I make calls out the PSTN the person on the other end hears it. If I use a Cisco ATA with an analog phone and call the same person again the static goes away, so I believe it is phone related. 2) When I call a non-voip phone when I stop talking (ie at the end of every sentence) the person on the other end hears some feedback/buzzing for a moment. Is anyone else using this phone and experiencing these issues? Has anyone else tried the GXP-2000 and decided to buy a different VoIP that they were impressed with (without spending too much money)? We have one GXP-2000 in house, and are trying to decided what phone to standardize on before we start rolling out them out to the users. Thanks, Aaron ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP-2000 registration issues
Greetings all I was wondering if any of you have found this problem. I have a setup with 80 GXP-2000s which work great until I decide to stop now my asterisk server. When it comes back up half of the phones (not always the same ones) fail to register with Asterisk. I set up the phones to register and send keep alives (register every 4 minutes, send keep alives every 20 seconds) to no avail. This is driving me nuts and Grandstream just gave me a generic answer. I'm using 1.0.1.9 firmware as the new one 1.0.1.12 is seriously flawed and buggy (The phones start degrading after a while, calls start to stutter and ringing starts becoming less and less loud until barely a beep). Thanks for your help. Jesus Mogollon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP 2000 Firmware 1.0.1.2
Greetings all Grandstream released a new firmware and it seems like the speaker phone problem has been fixed. However we updated to firmware 1.0.1.12 to fix the echo problem but found other problems were now created. The worst of these new problems is that the whole phone starts degrading, the volume starts getting lower and lower. The ringing starts fading and the calls start stuttering. The only way this can be fixed is by rebooting the phone. We were able to replicate this problem in all phones while some Polycoms we have do not suffer from this problem. Again, this problem happened AFTER we upgraded to the new firmware. Has anyone seen this? Jesus Mogollon Global IP Systems, LLC http://www.globalipsystems.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] GXP 2000 Firmware 1.0.1.2
Hi Lee: NTP is working as expected, but it does take a couple of minutes (!) to get the date from the server Jesus Mogollon 2005/8/25, Lee Archer [EMAIL PROTECTED]: Hi, do you have an on-site NTP server? I found that after the firmware update NTP from the * server stopped working. Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jesus MogollonSent: 24 August 2005 22:11To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] GXP 2000 Firmware 1.0.1.2 Greetings all Grandstream released a new firmware and it seems like the speaker phone problem has been fixed. However we updated to firmware 1.0.1.12 to fix the echo problem but found other problems were now created. The worst of these new problems is that the whole phone starts degrading, the volume starts getting lower and lower. The ringing starts fading and the calls start stuttering. The only way this can be fixed is by rebooting the phone. We were able to replicate this problem in all phones while some Polycoms we have do not suffer from this problem. Again, this problem happened AFTER we upgraded to the new firmware. Has anyone seen this?Jesus MogollonGlobal IP Systems, LLChttp://www.globalipsystems.com/### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 501 Do Not Disturb issue
Greetings all! I bought some Polycom 501s and got them installed, configured and running. I'm using bootrom 2.6.2 and sip 1.5.2. My problem is that when I press the Do Not Disturb button, the phone stops responding and reboots by itself eventually. Has anyone seen this problem before? Jesus Mogollon ___ Asterisk-Users mailing list Asterisk-Users@lists.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] SMS on my own possible?
Asterisk + Kannel. When you need to send a message, call the kannel directive with a System call. This is assuming you can connect to a SMSC via SMPP. Jesus Mogollon2005/7/13, Ronald_Wiplinger [EMAIL PROTECTED]: I am thinking of SMS and wonder if I can set-up with Asterisk a SMSC anduse SMS to / from VoIP phones.Can anybody give me a hint? Or has anybody done that?byeRonald___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://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] Saydigits
Hi... has anyone written or seen a variation of Saydigits that behaves like Background (listening and responding to DTMF)? If there's such a beast, I'd sure like to know... if not, how hard would it be to implement? Jesus ___ Asterisk-Users mailing list Asterisk-Users@lists.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: Good soft-phone on Linux
How about Xlite for Linux? It's already out... www.xten.com2005/6/27, Hamish Whittal [EMAIL PROTECTED]: Hi Folks,I am wanting advise on a good soft-phone on Linux. I have looked at Gnophone but cannot seem to get it to compile under debian sarge. I amnow looing at sipXphone seem to be picking up that it is not thatstable, but perhaps someone here can advise on what softphone I can useon Linux. Thanks in advance,Hamish ---| Hamish Whittal| Mobile: +27 82 803 5533 || QED Technologies cc | landline: +27 21 671 7710 || 21 Marne Avenue, Claremont, Cape Town | fax:+27 21 674 9184 ||fortune cookie below autogenerated_|You are confused; but this is your normal state. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Gnudialer
Hi there! Is there anyone out there using gnudialer? I tried vicidial but couldn't get it to work (does vicidial support SIP trunks anyways?). Gnudialer seems to be simpler, though their web interface needs a little work (version 2.0 seems like a step in the right direction but it isn't out yet). How do you register an agent? The documentation is lacking... Jesus Mogollon ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Predictive Dialer
Hi all: Has anybody used ny of the predictive dialers for Asterisk? I got the assignment of installing one, so I'd like to get any opinions beforehand. I'll post my experiences once installed and up-and-running! Thanks! Jesus Mogollon ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Interface bonding + asterisk
Hi all I installed asterisk on a dual PIII 700 with two NICs. I then proceeded to configure both NICs with bonding enable (bonding miimon=100 mode=1). I know certain features (like load balancing) under a bonded configuration is not understood by some switches, so I configured it using mode=1 (Failover only). The problem I'm having is that, sometimes, calls start fine but then one of the parties loses audio (it could be the caller of the callee who loses audio, there is no pattern). I was wondering if someone has hit the same wall as me. There are people using this server right now, so I haven't tried the no-bonding option as it means downtime. Any help would be appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web interface for realtime Mysql friends/peer
So am I (sorry to drop in like this). I'm a programmer and I'm open to start a project like this based on this attempt. Let me know.On Apr 8, 2005 9:46 AM, Douglas Conrad [EMAIL PROTECTED] wrote:Spencer,I am interested for your asterisk manager.Can you send for me?[]sDouglas ConradG.Marshall escreveu:Hello,It was written to manage asterisk in a postgres database, not MySQL.Itwas written to add sip_users, sip_peers, dialplans etc.If you are stillinterested, I will send you the php.As I have written, it is for postgres, not MySQL.SpencerMarshall,I am interested in seeing what you wrote to manage MySQL databaseobjects.By the way, latest version of OpenOffice comes with a MySQLAdministrator GUI to manage tables and data. This is something to lookat too.Seshu Kanuri-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of G.MarshallSent: Wednesday, April 06, 2005 2:27 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Web interface for realtime Mysqlfriends/peerThanksButI was looking for a more complete solution like areski or astccI found nothing so I wrote my own, but they are for postgres.They arenot complete by no means.If you are interested, I will let you have alook at what I have done, and if you provide constructive critisism, Iwill be happy to release the php under the same licence as Asterisk.LaurentAt 11:12 06/04/2005 +0200, Matteo Brancaleoni wrote:phpmyadmin :)Matteo.Il giorno mer, 06-04-2005 alle 20:05 +1100, Laurent Foulonneau hascritto:Hello list,Does anyone know about a web/php interface to deal with users inRealtime'sMysql database (sipusers and sippeers tables) ?Thanks in advanceLaurent___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005--No virus found in this outgoing message.Checked by AVG Anti-Virus.Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersNOTICE: If received in error, please destroy and notify sender.Senderdoes not waive confidentiality or privilege, and use is prohibited.___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] dialparties.agi
Greetings. Has anyone used dialparties.agi (part of AMP)?. In my setup, every once in a while, the script monopolizes the CPU driving it to a 100% usage and won't terminate. has anyone seen this problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
