Re: [Asterisk-Users] Digium G.729 vs. IPP G.729
http://lists.digium.com/pipermail/asterisk-dev/2004-September/006163.html - Original Message - From: Boris Bakchiev To: asterisk-users@lists.digium.com Sent: Monday, April 18, 2005 1:31 PM Subject: [Asterisk-Users] Digium G.729 vs. IPP G.729 Hi, Did anyone compare G.729 implementations (from Digium and the =ne based on IPP) on features, stability, quality and =eliabilty? It would be intresting to know how they fair against each =ther. I could be wrong, but in my testing I did notice a bit more hiss =n Digiums codec thein IPPs. Anyone? Internet communications cannot =e guaranteed to be secured or error-free as information could be =ntercepted, corrupted, lost, destroyed, arrive late or incomplete, or =ontain viruses. Therefore, we do not accept responsibility for any =rrors or omissions that are present in this message, or any attachment, =hat have arisen as a result of e-mail transmission. If verification is =equired, please request a hard-copy version. Any views or opinions =resented are solely those of the author and do not necessarily =epresent those of the company. ___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-user ___ 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] Dynamic Dialplan - Turn VM on/off?
Rod,
Here is my macro for this:
[macro-sipexten]
exten = a,1,VoicemailMain(${ARG1})
exten = a,2,Hangup()
exten = s,1,DBGet(NATIMEOUT=Features/${EXTEN}/NATIMEOUT)
exten = s,2,Dial(${ARG2},${NATIMEOUT})
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s,102,Goto(s,350)
exten = s,350,SetVar(NATIMEOUT=30)
exten = s,351,Goto(s,2)
As you can see it picks it up from DB with default being 30secs if no DB
entry exist.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Monday, 18 April 2005 15:58
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Dynamic Dialplan - Turn VM on/off?
G'day. I've been working with * for some time now, but mostly from a
enterprise perspective. I've just setup my own box at home and want to
enable some more home user type functionality.
Does anyone have a trick to allow the dynamic modification of the
dialplan by users? I want the ability to switch voicemail on/off (or
at
least alter the timeout).
In essence, I want to simulate the act of manually turning an
answering
machine on when you leave home (for my wife).
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Re: [Asterisk-Users] dynamic callrouting and billing?
I assume you'll be using IAX2 to connect all the servers? In each case, all you need is to match the pattern for the extension then send the call to another * server for final processing. If you only want to maintain this in one place, you could use ARA (Asterisk Realtime Architecture) and store the dialplan in a central database. I've tested this, and it (the dialplan part of ARA) seems to work OK. Given that the call routing will only be 30 lines per server config, I'd probably just manage them in the traditional (distributed, text file based) sense myself. As far as billing goes, we're writing our own system to use the asterisk CDR (stored locally on each server). We haven't determined a roll-up strategy for the databases yet, though being SQL, this is pretty easy to handle. - Original Message - From: maka [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, April 18, 2005 3:35 PM Subject: [Asterisk-Users] dynamic callrouting and billing? Hi everyone, I am trying to figure out a plan for dynamic call forwarding between multiple asterisk servers. I would be dealing with around 30 different extension prefixes, each handled by a distinct asterisk server. Is there a sort of dynamic call routing feature to accomplish this, or I would have to statically describe each extension prefix in extensions.conf (not that it's too much to do any way, but it would be better done dynamically) ? Also, is anyone aware of a free centralized billing solution that I can take a look at so I could possibly start working on my own? Cheers ___ 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] hangs pc
This could be any one of about 1.32 million things. Did the PC work OK before you put RH9/Asterisk on it? What sort of BRI card is it? Have you tested the card under another application/OS/platform? What version of Asterisk are you running? Is the BRI card sharing interrupts with anything else? What version of Libpri? What version of Zaptel drivers? What did you eat for dinner last night? What is your favourite sporting team? What is the average velocity of a swallow? A little more information may be helpful. - Original Message - From: Altus Snyman [EMAIL PROTECTED] To: asterisk asterisk-users@lists.digium.com Sent: Monday, April 18, 2005 3:29 PM Subject: [Asterisk-Users] hangs pc Good day all I installed asterisk on a pc with redhat 9 and a 4port bri eachtime a call comes in,iax,sip,pstn it just hangs the pc Top shows 75% of the cpu goes to asterisk? Any Idea why? Please Help ___ 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] Dynamic Dialplan - Turn VM on/off?
Thanks Boris. I think I can follow that logic!
- Original Message -
From: Boris Bakchiev [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 18, 2005 4:17 PM
Subject: RE: [Asterisk-Users] Dynamic Dialplan - Turn VM on/off?
Rod,
Here is my macro for this:
[macro-sipexten]
exten = a,1,VoicemailMain(${ARG1})
exten = a,2,Hangup()
exten = s,1,DBGet(NATIMEOUT=Features/${EXTEN}/NATIMEOUT)
exten = s,2,Dial(${ARG2},${NATIMEOUT})
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s,102,Goto(s,350)
exten = s,350,SetVar(NATIMEOUT=30)
exten = s,351,Goto(s,2)
As you can see it picks it up from DB with default being 30secs if no DB
entry exist.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Monday, 18 April 2005 15:58
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Dynamic Dialplan - Turn VM on/off?
G'day. I've been working with * for some time now, but mostly from a
enterprise perspective. I've just setup my own box at home and want to
enable some more home user type functionality.
Does anyone have a trick to allow the dynamic modification of the
dialplan by users? I want the ability to switch voicemail on/off (or
at
least alter the timeout).
In essence, I want to simulate the act of manually turning an
answering
machine on when you leave home (for my wife).
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This message (and any associated files) is intended only for the use of =he
individual or entity to which it is addressed and may contain =nformation
that is confidential, subject to copyright or constitutes a =rade secret. If
you are not the intended recipient you are hereby =otified that any
dissemination, copying or distribution of this =essage, or files associated
with this message, is strictly prohibited. =f you have received this message
in error, please notify us immediately =y replying to the message and
deleting it from your computer. Messages =ent to and from us may be
monitored...
Internet communications cannot be guaranteed to be secured or error-free =s
information could be intercepted, corrupted, lost, destroyed, arrive =ate or
incomplete, or contain viruses. Therefore, we do not accept =esponsibility
for any errors or omissions that are present in this =essage, or any
attachment, that have arisen as a result of e-mail =ransmission. If
verification is required, please request a hard-copy =ersion. Any views or
opinions presented are solely those of the author =nd do not necessarily
represent those of the company.
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[Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting
Jesse Guardiani [EMAIL PROTECTED] writes: I don't know about X-Lite, but sjphone seems only to support OSS. One of my requirements is ALSA support. Thus linphone and gnomemeeting. But, interestingly, gnomemeeting seems to be the only client capable of full duplex audio using ALSA+DMIX+DSNOOP+ASYM. Ah, I remember a thread about that on the GM list a couple of weeks ago, so that was you I presume. Well, XLite is OSS too, afaik, so that probably wouldn't help you either. Anway, pushing for an GM alpha snapshot with SIP support might still be an option compared to going through the H323 pile. Damien promised me twice http://mail.gnome.org/archives/gnomemeeting-list/2005-February/msg00018.html http://mail.gnome.org/archives/gnomemeeting-list/2005-April/msg00069.html to produce something workable, i.e. a release 1.3.1 as per the last mail. So if you reminded him too that at least some people are waiting for GM SIP support, it might accelerate the process a bit :) Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting
Bruno Hertz wrote: Jesse Guardiani [EMAIL PROTECTED] writes: I don't know about X-Lite, but sjphone seems only to support OSS. One of my requirements is ALSA support. Thus linphone and gnomemeeting. But, interestingly, gnomemeeting seems to be the only client capable of full duplex audio using ALSA+DMIX+DSNOOP+ASYM. Ah, I remember a thread about that on the GM list a couple of weeks ago, so that was you I presume. Well, XLite is OSS too, afaik, so that probably wouldn't help you either. xlite works OK with the OSS emulation for Alsa. B. ___ 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] dynamic callrouting and billing?
On Mon, 2005-04-18 at 16:18 +1000, Rod Bacon wrote: I assume you'll be using IAX2 to connect all the servers? In each case, all you need is to match the pattern for the extension then send the call to another * server for final processing. If you only want to maintain this in one place, you could use ARA (Asterisk Realtime Architecture) and store the dialplan in a central database. I've tested this, and it (the dialplan part of ARA) seems to work OK. Given that the call routing will only be 30 lines per server config, I'd probably just manage them in the traditional (distributed, text file based) sense myself. Isn't that what DuNDI (or whatever the correct capitalisation is), is supposed to do? My understanding (and I hope people will correct me if I am wrong) is that dundi is to asterisk what BGP4 is to routers... except that you never actually store the entire table locally, you ask your 'peers' at the time you want to dial... I've not yet had a chance to look into this yet, but it is on my todo list... Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] App_Conference
Anyone tried to build app_conference lately? I'm trying to setup a large conference where i speaker can talk to many listeners, for example 1 speaker and about 100 listeners (who can not speak back to the speaker, 1 way audio only) However, if i try to build app_conference against 1.0.6 or 1.0.7 it won't compile with an error message: make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o app_conference.o app_conference.c gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o conference.o conference.c conference.c: In function `create_conf': conference.c:614: warning: implicit declaration of function `__use_ast_pthread_create_instead__' gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o member.o member.c member.c: In function `member_exec': member.c:76: error: structure has no member named `cid' member.c:76: error: structure has no member named `cid' member.c:76: error: structure has no member named `cid' member.c:165: warning: unused variable `ignore_speex_count' make: *** [member.o] Error 1 _ Direct antwoord op je vragen: gebruik MSN Messenger http://messenger.msn.nl/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk PBX with X100P in India
+++ Min Hwan Chang [16/04/05 12:48 -0700]: Vikram, Would they really be able to tell if I have VOIP and POTS terminating on the PBX? Theoretically, its not like I'll be using this 100% of the time for sending VOIP calls to the POTS line. Probably maybe once or twice a month? It's main function is to act as a PBX with voicemail and such. Also regarding the setup of the X100P, would you be able to send me your extensions.conf? I have a feeling that its the setup of the extensions.conf which is bungling my attempts currently. In zapata.conf/zaptel.conf, are there any changes I need to make it work in India? I remember there being a setting for India, would I need to set that? Regards and much grateful thanks, Min I've used the X100P a lot here in India is works perfectly without any real config changes for incomming and outgoing calls. Are you sure you are on a real PSTN line I mean one from a Telco and not from an internal EPBX which would need you to dial a prefix to get a outside line. And if you are on an EPBX line and you already know what I mentioned aboove amybe your EPBX is giving some non standard tones or something there are many cheap EPBX's in India which really act wierd sometimes. On 4/15/05, Vikram Rangnekar [EMAIL PROTECTED] wrote: Just so that you know that would be considered illigal in India if you are planning to have VIP extensions on that Asterisk install also. Many people have been raided and even sent to jail for terminating PSTN and VOIP on the same PBX. -- regards Vikram (http://www.vicramresearch.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 Sorry I dont have that config anymore I switched to E1 but you should make sure u have the signalling right FXO cards have fxs signally and vice versa. also there is no special setting for India atall I've used the card here lots and also in the Us never done any country specific settings for it to work. Send me your zaptel and zapata configs and i'll check to see what u are doing wrong. If you want mail me offlist on [EMAIL PROTECTED] and if you are in bombay give me a call on 9819817434 I'll be glad to help. -- regards Vikram (http://www.vicramresearch.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] Re: asterisk + OH323 + NAT + gnomemeeting
Brian Capouch [EMAIL PROTECTED] writes: Bruno Hertz wrote: Jesse Guardiani [EMAIL PROTECTED] writes: I don't know about X-Lite, but sjphone seems only to support OSS. One of my requirements is ALSA support. Thus linphone and gnomemeeting. But, interestingly, gnomemeeting seems to be the only client capable of full duplex audio using ALSA+DMIX+DSNOOP+ASYM. Ah, I remember a thread about that on the GM list a couple of weeks ago, so that was you I presume. Well, XLite is OSS too, afaik, so that probably wouldn't help you either. xlite works OK with the OSS emulation for Alsa. Sure. I felt though the main trouble spot was asym properly working with the OSS emu (dmix and dsnoop apparently do). If you could confirm it does, all OSS only softphones would of course be candidates given the above requirements. Regards, Bruno. ___ 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] SIP/iax devices in Russia
Yes, sipuras work well in Russia. Actually, they're so configurable that I think they'll work everywhere. You'll need to re-configure to make them detect/generate Russian tone standard. snacktime wrote: Will sip/iax devices designed for European use also work in Russia? I'm specifically looking at using the Sipura ata's if anyone can confirm they work. Chris ___ 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 begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: IPswitch: How to use speed dialing?
tgj wrote:
Hi Ronald,
I must admit I am getting confused now.
I understand that you have a problem getting Speed Dial Buttons to work. The
problem as I understand it is that the calls are placed in the wrong
context.
To solve that problem I have asked you to make sure that you have typed a
valid context on the configuration page. Have you tried that?
I think thats all you need to do, how do I post an example of that? It's a
fairly easy thing to do.
Thorben
What is the right syntax to do that?
Context for dialing a trunk line is trunkint
Peter has the phone number 011-234-5678
How to set it up as a speed dial number? Below are all info you may need:
The phone 601 (= Monitor extension) is a Sip phone,
[general]
context=default; Default context for incoming calls
[601]
type=friend
username=601
secret=dont+tell+you
canreinvite=no
host=dynamic
dtmfmode=rfc2833
[EMAIL PROTECTED]
nat=yes
callgroup=1
pickupgroup=1
callerid=Ronald Hotline,601
qualify=1000
extensions.conf
[default]
...
include = trunkint
...
[trunkint]
;
; International long distance through trunk
; . other lines deleted
exten = _9011Z.,107,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9011Z.,108,hangup
bye
Ronald
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Re: [Asterisk-Users] Re: IPSwitchBoard Version 0.91 Released
tgj wrote: Thorben, I hope you find some time to make all more smoothly. It is a great product, but there are still some unclear things. 3. One IAX2 is simple to taken The three lines in Exensions / Extensions tab look like: IAX2 623 IAXy at home 623 Unspecified Internal (it is in the moment not connected) IAX2 NuFoneNuFone (Toll free USA) 6.225.202.72 Lines IAX2 demoDigium16.207.245.47 Main Extensions The button NuFone is always EMTPY (in Panel / Lines) 3. Have you got a button on the panel? (But with no text on it)? Yes, I got empty buttons there same for the zap lines/phones ___ 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] analog gsm router
Good day all I have a analog gsm router and a 4 port bri card:-) How do I get the gsm router to work with asterisk I tried adding a voicetronix card but the 2 cards doen not seem to work together,it gives a unresolved symbols error when starting up Any Ideas Please Can you add 2 zaptel device,different ones? Like the Junghannes and a diguim analog card? Please help and advice Thanks ALtus ___ 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] Got SIP response 302 Moved Temporarily back....
Hello everyone. How was your weekend? Anyway... 'Got SIP response 302 Moved Temporarily back from 192.168.10.24' Lately I've been getting this error... well i am at a loss as to why I am getting this when on Friday I was able to make a pass-through call with no problems. +--+ +-+ +---+ +-+ |Net2Phone |==|sip.Net2Phone.com||Asterisk(*)||SIP Phone| |MAX IP10 | +-+ +---+ |GS BT-100| +--+ (GateWay) +-+ [ip 196.x.x.x] [ip 66.33.157.12] [ip 165.x.x.x][ip 192.168.10.24] Asterisk Server(GateWay) has two eth cards - one with the external ip of 165.x.x.x via ppp0 and the other and internal ip of 192.x.x.x Now on Friday this setup worked 100% for a pass through - but now, I keep on getting this 302 error and I can't see how SIP is ending up in a HAIRPIN senario. DialPlan is simple: exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Dial(SIP/Receprion|20|tr) Asterisk(*) Output: -- Executing Answer(SIP/3828106029-29bb, ) in new stack -- Executing Wait(SIP/3828106029-29bb, 1) in new stack -- Executing Dial(SIP/3828106029-29bb, SIP/Reception|20|tr) in new stack -- Called Reception Apr 18 09:46:22 NOTICE[1841]: channel.c:1812 ast_set_write_format: Unable to find a path from slin to g723 Apr 18 09:46:22 WARNING[1841]: indications.c:78 playtones_alloc: Unable to set 'SIP/3828106029-29bb' to signed linear format (write) -- Got SIP response 302 Moved Temporarily back from 192.168.10.204 -- SIP/Reception-e6bf is busy == Everyone is busy/congested at this time (1:1/0/0) Any help on this issue will be apreciated. Thank you. Kindly, Etienne Pretorius ___ 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] App_Conference
I believe you need to modify a little bit member.c file in CVS version they use cid, but in stable version callerid. Just replace properly cid with callerid. It should help with that problem. For example: chan-cid.cid_num change to chan-callerid On Mon, 2005-04-18 at 10:04, E rikje wrote: Anyone tried to build app_conference lately? I'm trying to setup a large conference where i speaker can talk to many listeners, for example 1 speaker and about 100 listeners (who can not speak back to the speaker, 1 way audio only) However, if i try to build app_conference against 1.0.6 or 1.0.7 it won't compile with an error message: make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o app_conference.o app_conference.c gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o conference.o conference.c conference.c: In function `create_conf': conference.c:614: warning: implicit declaration of function `__use_ast_pthread_create_instead__' gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o member.o member.c member.c: In function `member_exec': member.c:76: error: structure has no member named `cid' member.c:76: error: structure has no member named `cid' member.c:76: error: structure has no member named `cid' member.c:165: warning: unused variable `ignore_speex_count' make: *** [member.o] Error 1 _ Direct antwoord op je vragen: gebruik MSN Messenger http://messenger.msn.nl/ ___ 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] analog gsm router
Hi, Can you add 2 zaptel device,different ones? Like the Junghannes and a diguim analog card? Please help and advice yes you can. use fxo port cards for this. Matteo. ___ 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] Got SIP response 302 Moved Temporarily back....
Got some debug info... please see attachement. Quoting [EMAIL PROTECTED]: Hello everyone. How was your weekend? Anyway... 'Got SIP response 302 Moved Temporarily back from 192.168.10.24' Lately I've been getting this error... well i am at a loss as to why I am getting this when on Friday I was able to make a pass-through call with no problems. +--+ +-+ +---+ +-+ |Net2Phone |==|sip.Net2Phone.com||Asterisk(*)||SIP Phone| |MAX IP10 | +-+ +---+ |GS BT-100| +--+ (GateWay) +-+ [ip 196.x.x.x] [ip 66.33.157.12] [ip 165.x.x.x][ip 192.168.10.24] Asterisk Server(GateWay) has two eth cards - one with the external ip of 165.x.x.x via ppp0 and the other and internal ip of 192.x.x.x Now on Friday this setup worked 100% for a pass through - but now, I keep on getting this 302 error and I can't see how SIP is ending up in a HAIRPIN senario. DialPlan is simple: exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Dial(SIP/Receprion|20|tr) Asterisk(*) Output: -- Executing Answer(SIP/3828106029-29bb, ) in new stack -- Executing Wait(SIP/3828106029-29bb, 1) in new stack -- Executing Dial(SIP/3828106029-29bb, SIP/Reception|20|tr) in new stack -- Called Reception Apr 18 09:46:22 NOTICE[1841]: channel.c:1812 ast_set_write_format: Unable to find a path from slin to g723 Apr 18 09:46:22 WARNING[1841]: indications.c:78 playtones_alloc: Unable to set 'SIP/3828106029-29bb' to signed linear format (write) -- Got SIP response 302 Moved Temporarily back from 192.168.10.204 -- SIP/Reception-e6bf is busy == Everyone is busy/congested at this time (1:1/0/0) Any help on this issue will be apreciated. Thank you. Kindly, Etienne Pretorius ___ 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 SIP Debugging Enabled for IP: 192.168.10.24:5060 -- Executing Answer(SIP/3828106029-8e32, ) in new stack -- Executing Wait(SIP/3828106029-8e32, 1) in new stack -- Executing Dial(SIP/3828106029-8e32, SIP/Reception|20|tr) in new stack We're at 192.168.10.1 port 14468 Answering/Requesting with root capability 0x1 (g723) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 10 lines Reliably Transmitting (no NAT) to 192.168.10.24:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK0bf1b64b From: asterisk sip:[EMAIL PROTECTED];tag=as4a953271 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: X-Lite release 1103m Date: Mon, 18 Apr 2005 09:11:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 214 v=0 o=root 2433 2433 IN IP4 192.168.10.1 s=session c=IN IP4 192.168.10.1 t=0 0 m=audio 14468 RTP/AVP 4 101 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called Reception Apr 18 11:11:22 NOTICE[2433]: channel.c:1812 ast_set_write_format: Unable to find a path from slin to g723 Apr 18 11:11:22 WARNING[2433]: indications.c:78 playtones_alloc: Unable to set 'SIP/3828106029-8e32' to signed linear format (write) adsl-test*CLI -- SIP read from 192.168.10.24:5060: SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK0bf1b64b From: asterisk sip:[EMAIL PROTECTED];tag=as4a953271 To: sip:[EMAIL PROTECTED];tag=6fe736daf4223205 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.5.18 Contact: sip:@192.168.10.1 Diversion: sip:[EMAIL PROTECTED];reason=unconditional Content-Length: 0 --- (10 headers 0 lines)--- -- Got SIP response 302 Moved Temporarily back from 192.168.10.24 Transmitting (no NAT) to 192.168.10.24:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK0bf1b64b From: asterisk sip:[EMAIL PROTECTED];tag=as4a953271 To: sip:[EMAIL PROTECTED];tag=6fe736daf4223205 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: X-Lite release 1103m Content-Length: 0 --- -- SIP/Reception-fe13 is busy == Everyone is busy/congested at this time (1:1/0/0) Destroying call '[EMAIL PROTECTED]'___ 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] Got SIP response 302 Moved Temporarily back....
Sorry- Solved my own problem. I was playing around with the GS BudgeTone 100 and had set up call forwarding on... -- SIP read from 192.168.10.24:5060: SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK0bf1b64b From: asterisk sip:[EMAIL PROTECTED];tag=as4a953271 To: sip:[EMAIL PROTECTED];tag=6fe736daf4223205 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.5.18 Contact: sip:@192.168.10.1 Diversion: sip:[EMAIL PROTECTED];reason=unconditional Content-Length: 0 The reason=unconditional, gave me an indication... Oh well. Sorry to post about silly mistakes like this. Sheepishly, Etienne Pretorius Quoting [EMAIL PROTECTED]: Got some debug info... please see attachement. Quoting [EMAIL PROTECTED]: Hello everyone. How was your weekend? Anyway... 'Got SIP response 302 Moved Temporarily back from 192.168.10.24' Lately I've been getting this error... well i am at a loss as to why I am getting this when on Friday I was able to make a pass-through call with no problems. +--+ +-+ +---+ +-+ |Net2Phone |==|sip.Net2Phone.com||Asterisk(*)||SIP Phone| |MAX IP10 | +-+ +---+ |GS BT-100| +--+ (GateWay) +-+ [ip 196.x.x.x] [ip 66.33.157.12] [ip 165.x.x.x][ip 192.168.10.24] Asterisk Server(GateWay) has two eth cards - one with the external ip of 165.x.x.x via ppp0 and the other and internal ip of 192.x.x.x Now on Friday this setup worked 100% for a pass through - but now, I keep on getting this 302 error and I can't see how SIP is ending up in a HAIRPIN senario. DialPlan is simple: exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Dial(SIP/Receprion|20|tr) Asterisk(*) Output: -- Executing Answer(SIP/3828106029-29bb, ) in new stack -- Executing Wait(SIP/3828106029-29bb, 1) in new stack -- Executing Dial(SIP/3828106029-29bb, SIP/Reception|20|tr) in new stack -- Called Reception Apr 18 09:46:22 NOTICE[1841]: channel.c:1812 ast_set_write_format: Unable to find a path from slin to g723 Apr 18 09:46:22 WARNING[1841]: indications.c:78 playtones_alloc: Unable to set 'SIP/3828106029-29bb' to signed linear format (write) -- Got SIP response 302 Moved Temporarily back from 192.168.10.204 -- SIP/Reception-e6bf is busy == Everyone is busy/congested at this time (1:1/0/0) Any help on this issue will be apreciated. Thank you. Kindly, Etienne Pretorius ___ 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] Distributed organizations - large scale public sector rollout
Hi List I am working with a pilot project for a Norwegian regional government to evaluate Asterisk for a large number of sites and users. The goal of the project is to have a unified VoIP-system to replace the disorganized collection of legacy PBX in use today. By distributed organization I mean an organization that consists of many, dispersed units, each with it's own existing telephony system, and with distinct number series. The goals of a unified system are several: - Lower traffic cost through a common backbone between sites and a common exit-point to the PSTN (either via IP or legacy lines). - Lower admin cost through unified, centralized management. - Added value through rollout of applications (voicemail, conferencing, IVR) that become globally available in the system. My main concern is manageability. From what I have seen of the available management tools there are none that address the needs of a distributed system. They all seems aimed at the SMB market, and don't leverage resources such as LDAP directories. Does anyone have any experience with management tools for Asterisk in a similar scenario? I am also very interrested in getting in touch with people working in similar projects. There is a large political element in rolling out Open Source telephony on such a scale, and having a network of similar projects could be of great value. I hope to be able to post to this list on our progress. Best regards -- Eivind Trondsen Wingnut Information Systems Norway ___ 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] Cisco 7970 startup problem
Hello all. I have a problem with Cisco 7970. At startup this device asks for CTL (Certificate Trust List) file, and startup process stops. I am even can't boot this device. Does anyone know how to avoid this problem? It is said that in older versions of SCCP dummy file with the name CTLSEPmac addr.tlv placed to TFTP can solve this problem, but in newer versions it doesn't work. Please help! ___ 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] Error on install of AMP
Hi list! When doing a new install of AMP I get this error: Configuring install for your environment../usr/src/AMP/apply_conf.sh: line 67: /usr/sbin/amportal: Permission denied OK Is this something I should be worried about? By the way, I have created some install scripts to download spandsp, asterisk, and AMP for a CentOS 4 box. It still needs some further work as I havent figured out yet how to do an unmpromted install of the mysql stuff (and include a password prompt) but other than that it seems to work fine. It will do most things automagically. Is there any place to post it publicly? I'm not a fancy coder so feel free to laugh at this feeble attempt to create an install script (but while you're at it improve the code) :) Thanks!! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can anyone send me sample config files for asterisk and X-Lite?
http://www.asteriskguru.com/xlite.html /Z Vaniah Voip wrote: Vamsi Pottangi wrote: It would be easier if you could get send us your sip.conf entry and confiuration made in x-lite Also, please let us know where exactly the problem is. Is it while registering the x-lite or during the call and the exact error messages. Cheers, ~Vamsi On 4/18/05, Abraham WEI [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I just want to make the simplest call in which an X-Lite calls another X-Lite via asterisk. Unfortunately I failed time and time again. If someone is kind enough to show me sample config files by which asterisk works well, it will help me a lot. Best regards, Abe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: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 mailto: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 It might be easier if you started with [EMAIL PROTECTED] ___ 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 signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Siemens optiPoint 420 phone and Asterisk
Dear Richard, On Fri, 15 Apr 2005, [EMAIL PROTECTED] wrote: The latest firmware for optipoint420 advance SIP seems to be version 4.0.22A, released for HiPath8000. thanks for this information. I've contacted my customer adviser at Siemens, he'll try to organize me this version. What siemens PBX do you use? It's a HiPath 3300 (Rack version) with the extension containing 4 ISDN ports to connect to *. I don't know... maybe it will work... We only have several OptiPoint400 and they work fine. The risk of making the phone unuseable by installing a wrong firmware seems too high for me, so I won't try that. Thanks for the help! Bye, Franz ___ 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] Unbelievable...
As only one individual, I thought their statements were very straight- forward and clear. Having worked as a senior manager in a technical organization, a large number of tehcnical people simply do not comprehend some words (or read other words into whatever they happen to be reading), or, jump to conclusions based on their technical knowledge that are unreasonable (contractually or otherwise). The wording is very obviously oriented toward those types, and I'd bet a fair amount they _still_ receive calls that are clearly answered on their web site. Regardless of what their web site says, they've provided me with the best service of the half dozen itsp's that I've worked with directly. And, I don't work for them or represent them. It's safe to assume that this particular company is pretty much functionally illiterate given the tone and tact of the rest of their comms. They won't be around long. On Apr 17, 2005, at 2:58 PM, Rusty Shackleford wrote: Unbelieavable, and utterly disgraceful. Anyone found responsible for establishing such a policy would quickly find their ass on the street in any organization that understands the first thing about customer service. One doesn't build or protect a business by threatening and bullying one's customers. If one is worried about the bad impression that complainers are giving about the operation, figure out WHY they are driven to such extremes and DO SOMETHING ABOUT IT. It isn't rocket surgery. The principles of running an effective customer service organization are well known and readily available to anyone. The mind boggles... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of snacktime Sent: Sunday, April 17, 2005 2:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Unbelievable... Sure sounds like a veiled threat to me. Post something they don't like and find your support ticket ignored or possibly your account closed? Oh well guess I won't be getting any support from livevoip anytime soon:) Straight from the network status page on their website... If you are working a trouble ticket with LiveVoip support and start posting to mailing lists or newsgroups you are just wasting your time. LiveVoip LLC will not respond to such postings which in many cases are done to push support teams. If anything it will slow your ticket or cause the case to be closed. Our techs work hard for you! They are not going to take abuse in any form. Posting to these lists is done by some as a way of trying to obtain faster support or vent frustrations. LiveVoip has a Zero interest in these actions and will respond per our Terms Conditions if required. Let our people help you. That is what they get paid for. Are they busy? Of course. Do they work long hours? Duh. Treat them nice and Say Thanks. You will get further by being part of solutions, not part of the problems. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.15 - Release Date: 04/16/2005 ___ 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 ---End of Original Message- ___ 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] Dynamic Dialplan - Turn VM on/off?
G'day. I've been working with * for some time now, but mostly from a
enterprise perspective. I've just setup my own box at home and want to
enable some more home user type functionality.
Does anyone have a trick to allow the dynamic modification of the
dialplan by users? I want the ability to switch voicemail on/off (or at
least alter the timeout).
In essence, I want to simulate the act of manually turning an answering
machine on when you leave home (for my wife).
Lots of different ways to do those things... here's one basic example:
; toggle the ivr by dialing this extension
exten = 3950,1,DBget(ISIVRON=FEAT/ivron) ; if success, step 2, else 102
exten = 3950,2,GotoIf(${ISIVRON} == yes?3:102)
exten = 3950,3,DBdel(FEAT/ivron)
exten = 3950,4,Background(npi-ivroff)
exten = 3950,5,Hangup
exten = 3950,102,DBput(FEAT/ivron=yes)
exten = 3950,103,Background(npi-ivron)
exten = 3950,104,Hangup
Then insert something in your dialplan statements to read the DBget
values, and branch as appropriate.
That example essentially turns an IVR on/off by dialing extn 3950.
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Re: [Asterisk-Users] Re: Siemens optiPoint 420 phone and Asterisk
Hi Franz, ok, can you please inform me (the list) if the Optipoint 420 with the firmware 4.0.22A work with Asterisk. If so i will try to contact our contact at Siemens and organize some Optipoint 420. chan_cornet, which would support the proprietary Siemens protocol, is notpart of the CVS tree yet (thanks to Steffen Koepf for writing this).BTW Have you additional information about Steffen's chan_cornet. Is there a beta version of the chan_cornet available for testing. As mentioned in my first post we use for the moment oh.323 and i'm very intersted to testit if possible. thx in advance...Franz Knipp [EMAIL PROTECTED] wrote: Dear Richard,On Fri, 15 Apr 2005, [EMAIL PROTECTED] wrote: The latest firmware for optipoint420 advance SIP seems to be version 4.0.22A, released for HiPath8000.thanks for this information. I've contacted my customer adviser atSiemens, he'll try to organize me this version. What siemens PBX do you use?It's a HiPath 3300 (Rack version) with the extension containing 4 ISDNports to connect to *. I don't know... maybe it will work... We only have several OptiPoint400 and they work fine.The risk of making the phone unuseable by installing a wrong firmwareseems too high for me, so I won't try that.Thanks for the help!Bye,Franz___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 Do you Yahoo!? Yahoo! Small Business - Try our new resources site! ___ 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] Still having broadvoice issues
Hi folks, I'm still having troubles with broadvoice. I can either make calls or receive calls but not both. It all depends upon how I setup the SIP stanza. Here's my incoming settings (these allow me to receive calls) register = 9738281625:PASSWORD:[EMAIL PROTECTED]/ [broadvoice] username=9738281625 type=peer secret=PASSWORD nat=yes insecure=very host=sip.broadvoice.com port=5060 fromuser=9738281625 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband context=inbound-analogue canreinvite=no authname=9738281625 qualify=1000 disallow=all allow=g726 allow=g729 callerid=Mark Phillips 9738953503 This next bit works only when I want to receive calls register = 9738281625:PASSWORD:[EMAIL PROTECTED]/ [broadvoice] username=9738281625 type=peer secret=PASSWORD nat=yes insecure=very host=proxy.dca.broadvoice.com port=5060 fromuser=9738281625 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband context=inbound-analogue canreinvite=no authname=9738281625 qualify=1000 disallow=all allow=g726 allow=g729 callerid=Mark Phillips 9738953503 As you can see the only difference is the host definition. Any ideas would be greatly appreciated. Thanks de Mark ___ 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] cannot dial two phones using zap
On Mon, Apr 18, 2005 at 10:02:48AM +0800, Eddie wrote: So the Panasonic extension dialed by Zap/3/206 command will ring and Zap/4/221 will not ring at all, even before ext 206 is picked up? Yes, exactly. Zap/4/221 won't ring at all. If you have two extensions numbered 211 212, why are you using 206 and 221 in your Dial command? 211 212 is plugged to asterisk, for dialing purpose. 206 221 is the extension I want to dial to. I would try this: 1. Make sure either extension will ring all by itself. Yes, they do ring all by itself. Okay, so we know that either one will work by itself. 2. Ring both at the same time, but put them in the other order in the Dial() command and see if that makes a difference. I've tried this: exten = 3,1,Dial(Zap/3/206,10) exten = 3,2,Wait(2) exten = 3,3,Dial(Zap/4/221,10) exten = 3,4,Hangup Zap/3/206 won't hangup / timeout. It just keep ringing and won't stop. :) What does the * log tell you? Go to the CLI, set verbose 3 and see what happens when you dial the above dialplan. 3. Rather than having: channel = 3,4 try channel = 3 channel = 4 just for fun. Tried this. No difference. I'm not surprised, I didn't think it would do anything... 4. I don't know much about that Panasonic PBX, but are you sure calling two lines at the exact same time isn't messing it up? Not sure. If I were you, I would try testing without the panasonic PBX to make sure that the FXOs and your zap settings are correct. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] Changing Codecs when dialing out...
Hello all,
For the g723.1 pass-through the incoming call works fine, I have been playing
around a bit and was wandering if you can dynamically change the channel and
the associated devices using the channel to change their codecs for the
outbound call.
I have the following setup in extensions.conf
exten = _9NXXNXX,1,SetVar(SIP_CODEC=g723.1)
exten = _9NXXNXX,n,Dial(SIP/net2phone/*72${EXTEN:1}) ;net2phone via
net2phone
exten = _9NXXNXX,n,SetVar(SIP_CODEC=ulaw)
This works fine if under sip.conf [general] the first codec is g723.1 but say I
would like the devices (GS BudgeTone 100) to first register with a diffrent
codec and when entering the dialplan to change to the appropiate codec.
The output is as follows CLI:
== Parsing '/etc/asterisk/sip_notify.conf': Not found (No such file or
directory)
-- Executing SetVar(SIP/Reception-fddb, SIP_CODEC=g723.1) in new stack
-- Executing Dial(SIP/Reception-fddb, SIP/net2phone/*72[edited_out]) in
new stack
-- Called net2phone/*72[edited_out]
-- SIP/net2phone-438d answered SIP/Reception-fddb
Apr 18 13:49:39 NOTICE[3318]: chan_sip.c:1995 sip_answer: Changing codec to
'g723.1' for this call because of ${SIP_CODEC) variable
-- Attempting native bridge of SIP/Reception-fddb and SIP/net2phone-438d
Apr 18 13:49:39 NOTICE[3318]: channel.c:1845 ast_set_read_format: Unable to find
a path from g723 to alaw
Apr 18 13:49:39 NOTICE[3318]: channel.c:1812 ast_set_write_format: Unable to
find a path from ulaw to g723
Apr 18 13:49:39 WARNING[3318]: channel.c:2251 ast_channel_make_compatible: No
path to translate from SIP/Reception-fddb(8) to SIP/net2phone-438d(1)
Apr 18 13:49:39 WARNING[3318]: channel.c:3064 ast_channel_bridge: Can't make
SIP/Reception-fddb and SIP/net2phone-438d compatible
Apr 18 13:49:39 WARNING[3318]: res_features.c:976 ast_bridge_call: Bridge failed
on channels SIP/Reception-fddb and SIP/net2phone-438d
== Spawn extension (sip, 9[edited_out], 2) exited non-zero on
'SIP/Reception-fddb'
-- Got SIP response 481 Call Leg/Transaction Does Not Exist back from
66.33.157.12
As you can see the GS BudgeTone 100 hasn't changed its codec when the channel
was set to g723.1. Is there a command that I must pass through to ask the GS
BudgeTone 100 to change its codec to g723.1?
Thank you.
Kindly,
Etienne Pretorius
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Re: [Asterisk-Users] ISDN BRI vs. VOIP DID's, is it worth it?
On Sun, Apr 17, 2005 at 01:50:56PM -0700, snacktime said: On 4/17/05, Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED] wrote: I have been trying a did company for a few days. I find the service decent, but sound quality only moderate. Rather than spending 35 or so for monthly with did, I am considering an isdn bri at this location. How much more stable and reliable is bri or pri versus a voip did service? I like the concept of a bri more, but I do not get cid generation. Would anyone suggest bri over voip where available? I must say, I prefer higher voice quality. If anyone finds bri to be worth it (at about 54/month plus usage) please let me know what you think. I'm kind of asking the same questions myself right now. I think it depends a lot on what you are planning on using voip for. I also think that you are going to see reliability go up and up over the next year or two, so you have to take that into account also as you plan your infrastructure. I think new installations should at least be voip capable. No matter what the usage is, BRI / PRI will be more reliable. VoIP to a generic providor will never be as reliable as a dedicated connection to your telco carrier of choice. Now whether you can live with the level of reliability is another story :-) The big problem with with VoIP is lack of QoS beyond your local network. Probably the best situation is to get your VoIP from your local ISP where QoS can be implemented end to end. Other current VoIP issues include spotty Fax support and flakey SIP / IAX support - these should be resolved in time, but they are a big problem now (as the volume of emails on this list related to providor problems shows.) As for QoS support on ther internet in general, well, I wouldn't hold my breath, and that is what is really needed to increase reliability / sound quality. Right now I would not rely on voip 100% for something business critical. Personally I'm looking at using voip but having adequate pstn access as a backup, with the incoming DID numbers being able to automatically route to the pstn in case of failure.I know I can do this if my numbers are 800 numbers, but I've still not found a way to do this with local number DID's, although I'm still looking. Reliability on incoming lines is a lot more difficult to deal with then outgoing. As long as you * server has connectivity, you could have 4-5 different providers in your dialplan and have it cascade down through them on failure. Wish it was that easy with DID's. True, if the providor is totally down you can fail over, but if the providor is up but not working well, you will have sound quality problems, dropped calls, etc. and there isn't a good way of handling this at the moment (could probably handle this via some new * code to score a providor during a call and drop them from the list if there are too many dropped packets, etc.) ___ 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] Distributed organizations - large scale public sector rollout
Eivind Most obvious solution is snmp. Using snmp you can collect statistics and provision your remote systems. However, SNMP is an enabler and not the full solution. You still need to write SMUX agents and develop application MIBS that allow you to get/store application specific data. To my knowledge Asterisk does not support any MIB reporting to date. You will need to extend asterisk with scripts and applications to provide you the data. Most of scripting tools like perl or php have good support for SNMP. HTH Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eivind Trondsen Sent: Monday, April 18, 2005 5:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Distributed organizations - large scale public sector rollout Hi List I am working with a pilot project for a Norwegian regional government to evaluate Asterisk for a large number of sites and users. The goal of the project is to have a unified VoIP-system to replace the disorganized collection of legacy PBX in use today. By distributed organization I mean an organization that consists of many, dispersed units, each with it's own existing telephony system, and with distinct number series. The goals of a unified system are several: - Lower traffic cost through a common backbone between sites and a common exit-point to the PSTN (either via IP or legacy lines). - Lower admin cost through unified, centralized management. - Added value through rollout of applications (voicemail, conferencing, IVR) that become globally available in the system. My main concern is manageability. From what I have seen of the available management tools there are none that address the needs of a distributed system. They all seems aimed at the SMB market, and don't leverage resources such as LDAP directories. Does anyone have any experience with management tools for Asterisk in a similar scenario? I am also very interrested in getting in touch with people working in similar projects. There is a large political element in rolling out Open Source telephony on such a scale, and having a network of similar projects could be of great value. I hope to be able to post to this list on our progress. Best regards -- Eivind Trondsen Wingnut Information Systems Norway ___ 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] Installing Asterisk@Home on VMware Workstation 4.5.2- build 8848
Sorry! Got it! All set. On Sun, 17 Apr 2005 15:54:37 -0400, SCollins [EMAIL PROTECTED] wrote: Just curious what syntax did you use to load the VMware tools on Fedora Core 3? Thanks, Sean On Sat, 16 Apr 2005 16:50:56 +0200, [EMAIL PROTECTED] wrote: I installed asterisk 1.0.7 successfully on VMware workstation with fedora 3 as guest. Of course without any hardware only pure asterisk. It works fine for testing. SCollins wrote: Newbie Question Has anybody installed [EMAIL PROTECTED] on VMware Workstation (w/ WMware Tools)successfully? Thanks, Sean ___ 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 -- http://www.1and1.com/?k_id=8358073 ___ 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] queue - transfer calls
Hello, I am setting up an ACD using *, but found a an issue that I am not being able to resolve, and this might impact our * implementation. We have a call center with 4 agents, which should receive calls from their queue. But we also have a "call center management" team which should be able to talk to end customers in case the first level call center is not able to solve the problem. There are two issues there: 1. The agent cannot use the soft-phone TRANSFER button.. she has to press the pound key to transfer. This is not a 'terrible' issue, since it is just a matter of educating agents. 2. Attended transfer: If the agent transfers the call to someone in the management team, the call is immediately transferred, and the agent is not able to talk to the manager before. Is there a way to allow an agent to talk to the management befora actually transferring, so that he can explain the issue in advance Thank you very much Dov ___ 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] Problem with Livevoip incoming context
Are you behind a firewall? If so, did you NAT an IP to your * machine with a port forward for yourIAX port? Have you done IAX2 debug? Help iax2 should get you the correct syntax. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris MasonSent: Thursday, April 14, 2005 7:20 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Problem with Livevoip incoming context Done all that, still doesnt work. I do have outgoing and incoming, just cant get the incoming to come through the livevoip context. Thanks Chris MasonUS Number: (646)722-0001 US Fax (815)301-9759Skype: netconcepts From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley SilerSent: Thursday, April 14, 2005 5:50 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Problem with Livevoip incoming context Should have in iax.conf. ;This registers you to them register=username:password@64.34.59.73 ;THis context serves to ID incoming, if you ahve a DID it shoudl come here [livevoip] type=user secret=mySecret host=64.34.59.73 callerid="Livevoip IAX User" context=livevoip-in ;This one is your outgoing... [ToLiveVoIP]username=usernametype=peersecret=YourSecrethost=64.34.59.73 As long as your Dial Plan refrerences these correctly, you should get both in and out with incoming registered to your livevoip. Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists)Sent: Thursday, April 14, 2005 2:39 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Problem with Livevoip incoming context I have a newly provisioned livevoip account which registers OK but the incoming calls are not being authenticated as livevoip and only work as the guest context: [livevoip] type=user secret=mySecret host=64.34.59.73 callerid="Livevoip IAX User" context=livevoip-in [guest] type=user callerid="Guest IAX User" context=guest-iax-in Any ideas? Chris Mason www.anguillaguide.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
Fw: [Asterisk-Users] Analogue phone transfering
Hi guys, Any other ideas on this one ? Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. - Original Message - From: David Wilson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Friday, April 15, 2005 3:45 PM Subject: Re: [Asterisk-Users] Analogue phone transfering Hi Eric, Thanks for your reply and guidance. I've tried that but unfortunately am still battling with the same problem. Any other ideas ? Thanks for your help so far. My zapata.conf: [channels] signalling=fxs_ks callprogress=no ;causes problems with calls not being established correctly context=incoming echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=800 ; Asterisk trains to the beginning of the call, number is in milliseconds ;echotraining=yes usecallerid=yes callerid=asreceived callwaiting=no usedistinctiveringdetection=no busydetect=yes busycount=8 adsi=no relaxdtmf=yes faxdetect=incoming channel=1-3 signalling=fxo_ks context=default relaxdtmf=yes ;threewaycalling=yes transfer=yes adsi=no usecallerid=no channel=4 ;rxgain=70.0 ;txgain=50.0 Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 15, 2005 3:21 PM Subject: Re: [Asterisk-Users] Analogue phone transfering David Wilson wrote: Hi guys, How are you keeping ? I have an analogue phone plugged into a Digium FXS Zap module on my TDM card. The phone works well except that I cannot seem to transfer calls using the flash key. I don't seem to get another dialtone as indicated in: http://www.voip-info.org/wiki-Asterisk+tips+zap+transfer Any ideas what I've done wrong ? This is my zapata.conf: [channels] ; For analogue phone signalling=fxo_ks context=default channel=4 relaxdtmf=yes threewaycalling=yes transfer=yes adsi=no usecallerid=no rxgain=70.0 txgain=50.0 In zapata.conf you set options and then APPLY the options to a channel. As you can see you are specifying the channel before most of your otions so they are never applied. Move your channel= line AFTER the options you want to set. You might want to remove your rxgain and txgain so you don't blow out your eardrums. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wcte11xp digium card
Hi, does anyone here tried using wcte11xp (e1) for R2 signaling. I need help because I cant make libsupertone, linunicall and libmfcr2 work. Im getting an error every time I issue the command make. Btw, the R2 variant is Philippine R2. Please help. Thanks. Angelo ___ 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] Changing Codecs when dialing out...
Hello all,
For the g723.1 pass-through the incoming call works fine, I have been playing
around a bit and was wandering if you can dynamically change the channel and
the associated devices using the channel to change their codecs for the
outbound call.
I have the following setup in extensions.conf
exten = _9NXXNXX,1,SetVar(SIP_CODEC=g723.1)
exten = _9NXXNXX,n,Dial(SIP/net2phone/*72${EXTEN:1}) ;net2phone via
net2phone
exten = _9NXXNXX,n,SetVar(SIP_CODEC=ulaw)
This works fine if under sip.conf [general] the first codec is g723.1 but say
I
would like the devices (GS BudgeTone 100) to first register with a diffrent
codec and when entering the dialplan to change to the appropiate codec.
The output is as follows CLI:
== Parsing '/etc/asterisk/sip_notify.conf': Not found (No such file or
directory)
-- Executing SetVar(SIP/Reception-fddb, SIP_CODEC=g723.1) in new
stack
-- Executing Dial(SIP/Reception-fddb, SIP/net2phone/*72[edited_out])
in
new stack
-- Called net2phone/*72[edited_out]
-- SIP/net2phone-438d answered SIP/Reception-fddb
Apr 18 13:49:39 NOTICE[3318]: chan_sip.c:1995 sip_answer: Changing codec to
'g723.1' for this call because of ${SIP_CODEC) variable
-- Attempting native bridge of SIP/Reception-fddb and SIP/net2phone-438d
Apr 18 13:49:39 NOTICE[3318]: channel.c:1845 ast_set_read_format: Unable to
find
a path from g723 to alaw
Apr 18 13:49:39 NOTICE[3318]: channel.c:1812 ast_set_write_format: Unable to
find a path from ulaw to g723
Apr 18 13:49:39 WARNING[3318]: channel.c:2251 ast_channel_make_compatible: No
path to translate from SIP/Reception-fddb(8) to SIP/net2phone-438d(1)
Apr 18 13:49:39 WARNING[3318]: channel.c:3064 ast_channel_bridge: Can't make
SIP/Reception-fddb and SIP/net2phone-438d compatible
Apr 18 13:49:39 WARNING[3318]: res_features.c:976 ast_bridge_call: Bridge
failed
on channels SIP/Reception-fddb and SIP/net2phone-438d
== Spawn extension (sip, 9[edited_out], 2) exited non-zero on
'SIP/Reception-fddb'
-- Got SIP response 481 Call Leg/Transaction Does Not Exist back from
66.33.157.12
As you can see the GS BudgeTone 100 hasn't changed its codec when the channel
was set to g723.1. Is there a command that I must pass through to ask the GS
BudgeTone 100 to change its codec to g723.1?
;---
I also wanted to ask say you would like to have the above example in a seperate
context...
extensions.conf
[sip]
exten = _9NXXNXX,1,Goto(net2phone_net2phone,${EXTEN:1},1) ;goto
net2phone_net2phone context...
[net2phone_net2phone]
exten = _NXXNXX,1,SetVar(SIP_CODEC=g723.1)
exten = _NXXNXX,n,Dial(SIP/net2phone/*72${EXTEN}) ;net2phone via
net2phone
exten = _NXXNXX,n,SetVar(SIP_CODEC=ulaw)
In the net2phone_net2phone context - how would I pass the dialed extension
format from the Goto statement as above?
Thank you.
Kindly,
Etienne Pretorius
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RE: [Asterisk-Users] Problem with Livevoip incoming context
No, it's in a datacenter. The IAX stuff is working, just not registering. I did debug it, all it says is "UNAUTHENTICATED" Chris Mason www.anguillaguide.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley SilerSent: Monday, April 18, 2005 9:20 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Problem with Livevoip incoming context Are you behind a firewall? If so, did you NAT an IP to your * machine with a port forward for yourIAX port? Have you done IAX2 debug? Help iax2 should get you the correct syntax. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris MasonSent: Thursday, April 14, 2005 7:20 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Problem with Livevoip incoming context Done all that, still doesnt work. I do have outgoing and incoming, just cant get the incoming to come through the livevoip context. Thanks Chris MasonUS Number: (646)722-0001 US Fax (815)301-9759Skype: netconcepts From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley SilerSent: Thursday, April 14, 2005 5:50 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Problem with Livevoip incoming context Should have in iax.conf. ;This registers you to them register=username:password@64.34.59.73 ;THis context serves to ID incoming, if you ahve a DID it shoudl come here [livevoip] type=user secret=mySecret host=64.34.59.73 callerid="Livevoip IAX User" context=livevoip-in ;This one is your outgoing... [ToLiveVoIP]username=usernametype=peersecret=YourSecrethost=64.34.59.73 As long as your Dial Plan refrerences these correctly, you should get both in and out with incoming registered to your livevoip. Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists)Sent: Thursday, April 14, 2005 2:39 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Problem with Livevoip incoming context I have a newly provisioned livevoip account which registers OK but the incoming calls are not being authenticated as livevoip and only work as the guest context: [livevoip] type=user secret=mySecret host=64.34.59.73 callerid="Livevoip IAX User" context=livevoip-in [guest] type=user callerid="Guest IAX User" context=guest-iax-in Any ideas? Chris Mason www.anguillaguide.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] OT: USB handsets / softphones
Not to ignore the fact that this is the cheapest and installtion free VOIP device that you can use for a real conversation, without bothering about the protocols. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Saturday, April 16, 2005 10:57 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] OT: USB handsets / softphones I agree, this is a fun device. It's a lot easier to use than a headset. The sound quality is excellent. Just don't turn up the volume too much or you will get a lot of echo. Echo is less of a problem with a good usb headset. It's a little quirky. All the sound from your pc gets routed to the phone. You can set x-lite to send ringing elsewhere. You have to load the driver that comes with the phone to be able to dial from the phone keypad. to dial a call, you press the dail button, dial the number, and press the dial button again. I can stand by the USB U2 Phone sold at http://www.eezeephone.com connected to a Firefly Third Party Version of the Softphone. This is one of the best combos I have ever used. Voice quality is phenomenal when using GSM or ILBC at just one end (better if on bothe ends) Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Friday, April 15, 2005 2:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] OT: USB handsets / softphones Here is just my personal opinion on the whole thing as I spent a good deal of time on this myself. In the end I had MUCH better results, and better sound quality moving to a Sipura SPA-1001 and a $14.99 cordless phone (with $12 rebate at Best Buy). Not only does it sound better, I don't have to walk around carrying my huge laptop. Full review of the SPA-1001 will be on GeekGazette tonight. Kerry http://geekgazette.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vahan Yerkanian Sent: Friday, April 15, 2005 11:09 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] OT: USB handsets / softphones Hi all, After googling around and searching both * and xten archives, I was still unable to find a working pair of softphone/usb *handset* that work with both keypad operating the softphones buttons *and* working incoming call ringer on the handset. I'm hoping that, while being OT for * discussion, someone else on this list had luck with finding a pair that works, preferably with xten's xlite/xpro. Any feedback is appreciated. regards, Vahan NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] 99% CPU - CVS 03.28.05
Hey Everyone, I've been running a version of the CVS without issue until late last week when suddenly Asterisk would randomly hit 99% CPU and stop registering my DIDs. If I stop Asterisk with a 'stop now' and restart Asterisk all is well... for a bit. So far I have deducted the following. Happens randomly during day and night - not at present times nor frequency Happens when no calls are present (it is a very low usage test box) If console is left open with high verbosity no errors are reported, CPU usage just climbs to 99% and the DIDs die - I only know because of Nagios and the DIDs ring busy. 'top' clearly lists Asterisk as the CPU hog. both 'uptime' and 'top' confirm the usage and the culprit. The server is at a data centre and is hardly used. It is only used for Asterisk. I have looked at all the other logs and cannot find any thing else creating entries - mail, messages, boot, anything. As I said the server does very little so it would be easy to see other entries. The Asterisk logs show nothing out of the ordinary. The machine does not have any digium hardware in it, it uses SIP for inbound and IAX for outbound. Basic calling card and voicemail functions. I can move to a newer CVS but that seems like new variables... I know this one was working and still works on a local test box using the providers. I am mainly looking to find the best way to see where Asterisk is getting stuck (some type of loop?) J ___ 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] Follow-me script - user changeable options
I have a company wants a pbx that has follow-me type rules, i.e., the user has a series of contact numbers comprising of home numbers, overseas numbers, cell phone numbers, and they are dialed in sequence. This is easy enough but the option they want that I am having trouble with is the ability for the user to dial a number and specify their extension/password to authenticate, then they are able to change the main umber to first try so that they control the routing of the call. The idea here is that when the user travels they can set what is their primary number to be reached at. Any ideas? Chris Mason NetConcepts (264) 4897-5670 Fax: (264) 497-8463 Int: (646)722-0001 Fax: (815)301-9759 Yahoo IM: [EMAIL PROTECTED] ___ 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] wcte11xp digium card
Hi, Il giorno lun, 18-04-2005 alle 21:28 +0800, Nathaniel Angelo A. Torres (247talk) ha scritto: Hi, does anyone here tried using wcte11xp (e1) for R2 signaling. I need help because I cant make libsupertone, linunicall and libmfcr2 work. Im getting an error every time I issue the command make. Btw, the R2 variant is Philippine R2. perhaps attaching the error can be of some help? while devs listening here are very good, none of them has divination powers, till now. Matteo. -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ 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] Cisco 7940
Title: Untitled Document Hello list, Could you tell me if you ever succeeded in configuring Cisco 7940 and chan_skinny. How ? (I cannot configure my phone, almost any submenu is unavailable) Thx. -- Thomas RULMONT Responsable Commercial Alterys SA T. +32 87 325939 T. +32 486 863216 E. [EMAIL PROTECTED] ___ 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] TDM card periodic buzz
On Apr 13, 2005, at 5:01 PM, Andrew Kohlsmith wrote: On April 13, 2005 03:42 pm, Trent Tuggle wrote: The symptom is a loud, brief buzz, almost exactly every 6 seconds, on the dot. It is only audible to remote parties, when I use an analog phone connected to my Digium TDM card. All other audio through my Asterisk box is fine, including SIP phones, music on hold, voicemail, etc. But when the TDM400P is bridged to the PSTN through my IAX2 provider, I get this repeating buzz! With it occurring, log in and type zttest and let it run for a minute and tell us the accuracy min/max/avg. Opened pseudo zap interface, measuring accuracy... 100.00% 100.00% 99.987793% 100.00% 99.987793% 100.00% 100.00% 99.987793% 100.00% 100.00% 99.987793% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 99.987793% 100.00% 99.987793% 100.00% 100.00% 99.987793% 99.987793% 100.00% 99.987793% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% --- Results after 109 passes --- Best: 100.00 -- Worst: 99.987793 What exactly does zttest test? ___ 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] queue - transfer calls
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dov Bigio Sent: Monday, April 18, 2005 9:16 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] queue - transfer calls Hello, I am setting up an ACD using *, but found a an issue that I am not being able to resolve, and this might impact our * implementation. We have a call center with 4 agents, which should receive calls from their queue. But we also have a call center management team which should be able to talk to end customers in case the first level call center is not able to solve the problem. There are two issues there: The agent cannot use the soft-phone TRANSFER button.. she has to press the pound key to transfer. This is not a 'terrible' issue, since it is just a matter of educating agents. This one can be fixed if you want by going with the paid xten pro software. It has a transfer button. Attended transfer: If the agent transfers the call to someone in the management team, the call is immediately transferred, and the agent is not able to talk to the manager before. Is there a way to allow an agent to talk to the management befora actually transferring, so that he can explain the issue in advance In stead of transferring to the next level support have your agents park the call to lets say 700 it should give you something like 701 then call the next agent tell them what the problem is and to pickup exten 701. Thank you very much Dov ___ 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] zap device detects hangup when phone switches from answer machine announcement to recording
Hi martin. Maybe setting callprogress=no and busydetect=no, or increment the busycount parameter. all in zapata.conf you can read more about these parameters in the wiki at voip-info.org best regards - moy On 4/16/05, Martin Renschler [EMAIL PROTECTED] wrote: Hi, I have a Panasonic Cordless phone and want to use the built-in answer machine instead of an asterisk voice mailbox. The problem is now that the answer machine plays the announcement and exactly when it wants to record, asterisk reports a zap Hangup. The caller never even hears the beep. Any idea what is going on, any parameter in the zap config files that would fix that? Thanks /Martin ___ 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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ 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] Indicating when other party has answered
Here in Sweden when I make a call through the regular POTS, I get an polarity reversal when the callee has lift his phone and answered. Now I've got an Adit 600 with 40 FXS channels and want to emulate an regular POTS. But the Adit doesn't seem to support polarity reversal. Is there other standards how to indicate to the caller that the callee has answered the call? How does it work in other countries? Thanks! -- Daniel Nyström ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Fax questions
Jesse Guardiani wrote:
Thank you for you time to help setting up fax.
I still have some questions.
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(3)
[fax]
exten = 2201,1,Macro(faxreceive)
exten = 2202,1,Macro(faxreceive)
exten = 2203,1,Macro(faxreceive)
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR} ${CALLERIDNUM}
${CALLERIDNAME})
has the sender dial the extensions 2201 ~ 2203 ?
You said it would automatically go to [fax] if in zapata.conf is set
faxdetect=both
If so, than we could use NO number, but s as extension, would that be
right?
NOTE: asterisk automatically jumps to the [fax] context if you are using
faxdetect in your zapata.conf
NOTE2: mailfax is a custom script I wrote. This is what it looks like:
-- START mailfax script --
#!/bin/sh
FAXFILE=$1
RECIPIENT=$2
FAXSENDER=$3
FAXID=`date +%j%H%M%S`
tempfoo=fax
TMPFILE=`mktemp /tmp/${tempfoo}XX`
TMPFILE_A=`mktemp /tmp/${tempfoo}XX`.pdf
/usr/bin/tiff2pdf -p letter ${FAXFILE} ${TMPFILE_A}
metasend -b -t $RECIPIENT -s Fax from $FAXSENDER \
-f ${TMPFILE} -m 'text/plain' -n \
-m 'application/pdf;name=fax'${FAXID}'.pdf' -f ${TMPFILE_A} \
-D 'PDF Fax Document'
rm ${TMPFILE}
rm ${TMPFILE_A}
-- END mailfax script --
I don't have found metasend on my system. Do you know where it is?
I based this config on the excellent information found at the following
website:
http://scottstuff.net/scott/archives/000152.html
At this web site I find also a line:
|*CLI database put extensionemail 2202 [EMAIL PROTECTED]
How does that fit together with|
macro-faxreceive,103 ???
This methode would cover the part of receiving a fax via a zap device, how to
exend it to receive a fax from a remote gateway via G711?
bye
Ronald
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[Asterisk-Users] Help compiling zaptel in Debian
During the zaptel configuration at the end of it there is this error: post-install tor2 /sbin/ztcfg post-install wcusb /sbin/ztcfg post-install wcfxo /sbin/ztcfg post-install ztdynamic /sbin/ztcfg post-install ztd-eth /sbin/ztcfg post-install wct1xxp /sbin/ztcfg post-install wct4xxp /sbin/ztcfg post-install wcte11xp /sbin/ztcfg if [ -d /etc/modutils ]; then \ /sbin/update-modules ; \ fi depmod: *** Unresolved symbols in /lib/modules/2.4.20-686/misc/ztd-eth.o [ `id -u` = 0 ] /sbin/depmod -a || : depmod: *** Unresolved symbols in /lib/modules/2.4.20-686/misc/ztd-eth.o [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample /etc/zaptel.conf I have previously made the make oldconfig and the make dep ... I'm using debian with 2.4.20 kernel... any ideas..? thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM card periodic buzz
On April 18, 2005 10:17 am, Trent Tuggle wrote: Opened pseudo zap interface, measuring accuracy... --- Results after 109 passes --- Best: 100.00 -- Worst: 99.987793 What exactly does zttest test? That's not terribly bad; Were you able to tell if the buzz occurrs when the timing drops down below 100%? What else is this box doing? -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Motherboard failure with 2 Digium TE405P cards
Hello, I have spend a long time trying to figure out exactly what is the problem with one of my Asterisk servers, it is the only one at any of our locations that has two Digium quad T1 cards in it with 7 T1s connected to it. Most of the rest of our Asterisk servers run identical hardware except that they only have a single TE405P board in them. Here's what seems to happen to this system starting 6 months ago: Take brand new Asus motherboard with P4 processor, 2GB RAM, SATA or SCSI drives and two TE405P Digium quad T1 boards. Hook up one local and one long distance T1, hook up 4 crossover PRIs to other telco equipment, hook up one channelbank. The system will run perfectly for about 5 weeks, then randomly the channel bank users will notice a weird audio cracking sound and the system will crash. Upon investigation the second TE405P card will have it's lights all off and on reboot they will not go back on again. After frantically switching the PCI slot that the lights-out card was in to a free slot the card works again and everything is happy again, but now no digium card will work in the other slot again. Another 5 weeks or so passes and again one of the Digium quad cards stops working. At this point I swap out the entire system(including quad cards) with another system that has been running for 6 months with no problem and put the malfunctioning system in production with a single quad card(which now has been running fine 4 months later) and after 6 weeks it happens to the new system. The whole process repeats itself and I am now on my 3rd set of completely different components serving in this role(even with different brands of components) and my first PCI slot just failed last week. We need to have the capability to handle 7 T1s on this machine and it is not over-heated or overloaded from a system load standpoint. We also have $200 550W Enermax power supplies in these servers that have never failed us before. So here's the question, do two Digium TE405P boards draw too much power or do something else that would harm a brand new motherboard over time? Does anyone else out there run two quad board in production? if so what hardware do you use? I'm just looking for some user feedback before I contact Digium hardware support on this. Thanks, MATT--- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Unbelievable...
Rich Adamson [EMAIL PROTECTED] writes: As only one individual, I thought their statements were very straight- forward and clear. Having worked as a senior manager in a technical organization, a large number of tehcnical people simply do not comprehend some words (or read other words into whatever they happen to be reading), or, jump to conclusions based on their technical knowledge that are unreasonable (contractually or otherwise). The wording is very obviously oriented toward those types, and I'd bet a fair amount they _still_ receive calls that are clearly answered on their web site. Regardless of what their web site says, they've provided me with the best service of the half dozen itsp's that I've worked with directly. And, I don't work for them or represent them. Interesting you say that, since I thought their statement wasn't that offensive, but rather looked like a fairly emotional reaction to the severe pressure they might experience right now, and which, as they say, apparently starts comsuming resources better spent on trouble shooting. Especially, those of us who have already worked in some kind of online business will recognize the situation and mood they apparently are in, and how unpleasant it can be. Although, on the other hand, a pissed off customer understandably might have a hard time feeling compassionate. Anyway, I think that just because ppl take money for service doesn't necessarily obligate them to take any shit customers might come up with as well. It's the service which is paid for, so if it isn't delivered for whatever reasons, all one basically is entitled to is getting the money back and maybe compensation, depending on the type of service and contract. Also, it's clear whom they are addressing in that statement, i.e. those people who continue mounting pressure on them through various channels in a counterproductive and 'abusive' fashion, some of them maybe really just to 'vent frustrations'. Well, if so, why not let them do their venting in that particular direction as well and move on to the real issues ... ? Regards, Bruno. ___ 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] wcte11xp digium card
Hi Matteo, Please find attached excerpts of the error below: supertone.c:337: invalid type argument of `-' supertone.c:337: syntax error before xmlChar supertone.c: At top level: supertone.c:344: redefinition of `cur' supertone.c:263: `cur' previously defined here supertone.c:344: invalid type argument of `-' supertone.c:344: warning: data definition has no type or storage class supertone.c:345: syntax error before while supertone.c:357: syntax error before '' token supertone.c:357: warning: data definition has no type or storage class supertone.c:357: syntax error before '}' token supertone.c:357: conflicting declarations of `__result' supertone.c:357: `__result' previously declared here supertone.c:357: warning: `__result' was declared `extern' and later `static' supertone.c:357: `x' undeclared here (not in a function) supertone.c:357: `__s2' undeclared here (not in a function) supertone.c:357: syntax error before if supertone.c:357: conflicting declarations of `__result' supertone.c:357: `__result' previously defined here supertone.c:357: warning: data definition has no type or storage class supertone.c:357: syntax error before '}' token supertone.c:357: warning: data definition has no type or storage class supertone.c:357: syntax error before '}' token supertone.c:357: conflicting declarations of `__result' supertone.c:357: `__result' previously declared here supertone.c:357: warning: `__result' was declared `extern' and later `static' supertone.c:357: `__s1' undeclared here (not in a function) supertone.c:357: `set_id' undeclared here (not in a function) supertone.c:357: syntax error before if supertone.c:357: conflicting declarations of `__result' supertone.c:357: `__result' previously defined here supertone.c:357: warning: data definition has no type or storage class supertone.c:357: syntax error before '}' token supertone.c:359: conflicting types for `__retval' supertone.c:258: previous declaration of `__retval' supertone.c:359: `__len' undeclared here (not in a function) supertone.c:359: syntax error before if supertone.c:359: conflicting types for `__retval' supertone.c:359: previous declaration of `__retval' supertone.c:359: warning: data definition has no type or storage class supertone.c:359: syntax error before '}' token supertone.c:361: conflicting types for `__retval' supertone.c:359: previous declaration of `__retval' supertone.c:361: `__len' undeclared here (not in a function) supertone.c:361: syntax error before if supertone.c:361: conflicting types for `__retval' supertone.c:361: previous declaration of `__retval' supertone.c:361: warning: data definition has no type or storage class supertone.c:361: syntax error before '}' token supertone.c:363: warning: parameter names (without types) in function declaration supertone.c:363: conflicting types for `parse_tone_set' supertone.c:176: previous declaration of `parse_tone_set' supertone.c:363: warning: data definition has no type or storage class supertone.c:364: warning: parameter names (without types) in function declaration supertone.c:364: warning: data definition has no type or storage class supertone.c:365: syntax error before return supertone.c:372: redefinition of `cur' supertone.c:344: `cur' previously defined here supertone.c:372: invalid type argument of `-' supertone.c:372: warning: data definition has no type or storage class supertone.c:373: syntax error before '}' token supertone.c:375: warning: parameter names (without types) in function declaration supertone.c:375: warning: data definition has no type or storage class supertone.c:376: syntax error before '-' token supertone.c:376: conflicting types for `free' /usr/include/stdlib.h:569: previous declaration of `free' supertone.c:376: warning: data definition has no type or storage class supertone.c:208: register name not specified for `__result' supertone.c:208: register name not specified for `__result' supertone.c:226: register name not specified for `__result' supertone.c:226: register name not specified for `__result' supertone.c:238: register name not specified for `__result' supertone.c:238: register name not specified for `__result' supertone.c:250: register name not specified for `__result' supertone.c:250: register name not specified for `__result' supertone.c:357: register name not specified for `__result' supertone.c:357: register name not specified for `__result' make[1]: *** [supertone.lo] Error 1 make[1]: Leaving directory `/usr/src/libsupertone-0.0.2' make: *** [all] Error 2 Thank you for the help. Cheers, Angelo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matteo Brancaleoni Sent: Monday, April 18, 2005 10:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] wcte11xp digium card Hi, Il giorno lun, 18-04-2005 alle 21:28 +0800, Nathaniel Angelo A. Torres (247talk) ha scritto: Hi, does anyone here tried using wcte11xp (e1) for R2 signaling. I need help because I
RE: [Asterisk-Users] 99% CPU - CVS 03.28.05
I've heard this problem could be caused by the hold music. I forgot the name of the process mpeg or wavmpeg, something along those lines... - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moody Sent: Monday, April 18, 2005 9:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 99% CPU - CVS 03.28.05 Hey Everyone, I've been running a version of the CVS without issue until late last week when suddenly Asterisk would randomly hit 99% CPU and stop registering my DIDs. If I stop Asterisk with a 'stop now' and restart Asterisk all is well... for a bit. So far I have deducted the following. Happens randomly during day and night - not at present times nor frequency Happens when no calls are present (it is a very low usage test box) If console is left open with high verbosity no errors are reported, CPU usage just climbs to 99% and the DIDs die - I only know because of Nagios and the DIDs ring busy. 'top' clearly lists Asterisk as the CPU hog. both 'uptime' and 'top' confirm the usage and the culprit. The server is at a data centre and is hardly used. It is only used for Asterisk. I have looked at all the other logs and cannot find any thing else creating entries - mail, messages, boot, anything. As I said the server does very little so it would be easy to see other entries. The Asterisk logs show nothing out of the ordinary. The machine does not have any digium hardware in it, it uses SIP for inbound and IAX for outbound. Basic calling card and voicemail functions. I can move to a newer CVS but that seems like new variables... I know this one was working and still works on a local test box using the providers. I am mainly looking to find the best way to see where Asterisk is getting stuck (some type of loop?) J ___ 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] 99% CPU - CVS 03.28.05
Are you running any AGI scripts? On 4/18/05, Moody [EMAIL PROTECTED] wrote: Hey Everyone, I've been running a version of the CVS without issue until late last week when suddenly Asterisk would randomly hit 99% CPU and stop registering my DIDs. If I stop Asterisk with a 'stop now' and restart Asterisk all is well... for a bit. So far I have deducted the following. Happens randomly during day and night - not at present times nor frequency Happens when no calls are present (it is a very low usage test box) If console is left open with high verbosity no errors are reported, CPU usage just climbs to 99% and the DIDs die - I only know because of Nagios and the DIDs ring busy. 'top' clearly lists Asterisk as the CPU hog. both 'uptime' and 'top' confirm the usage and the culprit. The server is at a data centre and is hardly used. It is only used for Asterisk. I have looked at all the other logs and cannot find any thing else creating entries - mail, messages, boot, anything. As I said the server does very little so it would be easy to see other entries. The Asterisk logs show nothing out of the ordinary. The machine does not have any digium hardware in it, it uses SIP for inbound and IAX for outbound. Basic calling card and voicemail functions. I can move to a newer CVS but that seems like new variables... I know this one was working and still works on a local test box using the providers. I am mainly looking to find the best way to see where Asterisk is getting stuck (some type of loop?) J ___ 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 -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] There are 10 types of people. Those who understand binary, and those who don't. ___ 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] Help compiling zaptel in Debian
try auto-apt for getting dependencies satisfied on the fly while compiling. Manuel Casal wrote: During the zaptel configuration at the end of it there is this error: ___ 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] Motherboard failure with 2 Digium TE405P cards
Check the telco equipment you are plugging into (PBXes) with the crossovers.. Unless they are all on the same power grid and protected I would blame them. my two cents... On 4/18/05, mattf [EMAIL PROTECTED] wrote: Hello, I have spend a long time trying to figure out exactly what is the problem with one of my Asterisk servers, it is the only one at any of our locations that has two Digium quad T1 cards in it with 7 T1s connected to it. Most of the rest of our Asterisk servers run identical hardware except that they only have a single TE405P board in them. Here's what seems to happen to this system starting 6 months ago: Take brand new Asus motherboard with P4 processor, 2GB RAM, SATA or SCSI drives and two TE405P Digium quad T1 boards. Hook up one local and one long distance T1, hook up 4 crossover PRIs to other telco equipment, hook up one channelbank. The system will run perfectly for about 5 weeks, then randomly the channel bank users will notice a weird audio cracking sound and the system will crash. Upon investigation the second TE405P card will have it's lights all off and on reboot they will not go back on again. After frantically switching the PCI slot that the lights-out card was in to a free slot the card works again and everything is happy again, but now no digium card will work in the other slot again. Another 5 weeks or so passes and again one of the Digium quad cards stops working. At this point I swap out the entire system(including quad cards) with another system that has been running for 6 months with no problem and put the malfunctioning system in production with a single quad card(which now has been running fine 4 months later) and after 6 weeks it happens to the new system. The whole process repeats itself and I am now on my 3rd set of completely different components serving in this role(even with different brands of components) and my first PCI slot just failed last week. We need to have the capability to handle 7 T1s on this machine and it is not over-heated or overloaded from a system load standpoint. We also have $200 550W Enermax power supplies in these servers that have never failed us before. So here's the question, do two Digium TE405P boards draw too much power or do something else that would harm a brand new motherboard over time? Does anyone else out there run two quad board in production? if so what hardware do you use? I'm just looking for some user feedback before I contact Digium hardware support on this. Thanks, MATT--- ___ 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 -- Andrew Latham http://www.lathama.com [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] If any of the above are not working, we have bigger problems than my email. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re:queue - transfer calls
Thanks Ariel. Your 2nd suggestions seems a good bypass for this problem... it might be helpful here, thanks! About the 1st one (using paid X-Ten software), I am using paid X-Pro, which does have a transfer button... but ifIuse this button instead of pound, the calls simply hangs up.. But I think that unfortunately, this is the expected behaviour! ThanksDov From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Dov BigioSent: Monday, April 18, 2005 9:16 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Hello,I am setting up an ACD using *, but found a an issue that I am not beingable to resolve, and this might impact our * implementation.We have a call center with 4 agents, which should receive calls from theirqueue. But we also have a "call center management" team which should be able to talk to end customers in case the first level call center is not able tosolve the problem.There are two issues there:1. The agent cannot use the soft-phone TRANSFER button.. she has topress the pound key to transfer. This is not a 'terrible' issue, since it isjust a matter of educating agents.This one can be fixed if you want by going with the paid xten pro software.It has a transfer button.2. Attended transfer: If the agent transfers the call to someone in themanagement team, the call is immediately transferred, and the agent is notable to talk to the manager before. Is there a way to allow an agent to talkto the management befora actually transferring, so that he can explain theissue in advanceIn stead of transferring to the next level support have your agents park thecall to lets say 700 it should give you something like 701 then call thenext agent tell them what the problem is and to pickup exten 701.Thank you very muchDov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Unbelievable...
As only one individual, I thought their statements were very straight- forward and clear. Having worked as a senior manager in a technical organization, a large number of tehcnical people simply do not comprehend some words (or read other words into whatever they happen to be reading), or, jump to conclusions based on their technical knowledge that are unreasonable (contractually or otherwise). The wording is very obviously oriented toward those types, and I'd bet a fair amount they _still_ receive calls that are clearly answered on their web site. Regardless of what their web site says, they've provided me with the best service of the half dozen itsp's that I've worked with directly. And, I don't work for them or represent them. Interesting you say that, since I thought their statement wasn't that offensive, but rather looked like a fairly emotional reaction to the severe pressure they might experience right now, and which, as they say, apparently starts comsuming resources better spent on trouble shooting. Especially, those of us who have already worked in some kind of online business will recognize the situation and mood they apparently are in, and how unpleasant it can be. Although, on the other hand, a pissed off customer understandably might have a hard time feeling compassionate. Anyway, I think that just because ppl take money for service doesn't necessarily obligate them to take any shit customers might come up with as well. It's the service which is paid for, so if it isn't delivered for whatever reasons, all one basically is entitled to is getting the money back and maybe compensation, depending on the type of service and contract. Also, it's clear whom they are addressing in that statement, i.e. those people who continue mounting pressure on them through various channels in a counterproductive and 'abusive' fashion, some of them maybe really just to 'vent frustrations'. Well, if so, why not let them do their venting in that particular direction as well and move on to the real issues ... ? Given the number of people on this list that don't understand how nat works, why their registration fails, etc, I can just about guess at the type of support calls/emails they get and the level of hand-holding they have to be asked for to implement a relatively simple asterisk link. I honestly feel the wording on their pages are right on base. Their name to fame is not orented around 1,000's of home-bodies trying to implement their first sip adapter. Their business is certainly targeted at the high-volume traffic, and as such, they deal with more technical types that think the problem is always at other end. Those that are offended by the wording are probably very thin skinned techies. ___ 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] Motherboard failure with 2 Digium TE405P car ds
For this particular server all telco equipment is in a climate controlled room kept at 66 degrees F and they are all on APC SmartUPS rackmount power battery backups, Also all of these connections had previously been connected to other Digium cards in the last year with no issues. MATT--- -Original Message- From: Andrew Latham [mailto:[EMAIL PROTECTED] Sent: Monday, April 18, 2005 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Motherboard failure with 2 Digium TE405P cards Check the telco equipment you are plugging into (PBXes) with the crossovers.. Unless they are all on the same power grid and protected I would blame them. my two cents... On 4/18/05, mattf [EMAIL PROTECTED] wrote: Hello, I have spend a long time trying to figure out exactly what is the problem with one of my Asterisk servers, it is the only one at any of our locations that has two Digium quad T1 cards in it with 7 T1s connected to it. Most of the rest of our Asterisk servers run identical hardware except that they only have a single TE405P board in them. Here's what seems to happen to this system starting 6 months ago: Take brand new Asus motherboard with P4 processor, 2GB RAM, SATA or SCSI drives and two TE405P Digium quad T1 boards. Hook up one local and one long distance T1, hook up 4 crossover PRIs to other telco equipment, hook up one channelbank. The system will run perfectly for about 5 weeks, then randomly the channel bank users will notice a weird audio cracking sound and the system will crash. Upon investigation the second TE405P card will have it's lights all off and on reboot they will not go back on again. After frantically switching the PCI slot that the lights-out card was in to a free slot the card works again and everything is happy again, but now no digium card will work in the other slot again. Another 5 weeks or so passes and again one of the Digium quad cards stops working. At this point I swap out the entire system(including quad cards) with another system that has been running for 6 months with no problem and put the malfunctioning system in production with a single quad card(which now has been running fine 4 months later) and after 6 weeks it happens to the new system. The whole process repeats itself and I am now on my 3rd set of completely different components serving in this role(even with different brands of components) and my first PCI slot just failed last week. We need to have the capability to handle 7 T1s on this machine and it is not over-heated or overloaded from a system load standpoint. We also have $200 550W Enermax power supplies in these servers that have never failed us before. So here's the question, do two Digium TE405P boards draw too much power or do something else that would harm a brand new motherboard over time? Does anyone else out there run two quad board in production? if so what hardware do you use? I'm just looking for some user feedback before I contact Digium hardware support on this. Thanks, MATT--- ___ 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 -- Andrew Latham http://www.lathama.com [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] If any of the above are not working, we have bigger problems than my email. ___ 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] 99% CPU - CVS 03.28.05
thanks for the help... I knew I missed some info... Music on hold.. I am not using any form of it. As for AGIs... I do have AreskiCC installed but it is used for only some calls. I discounted it as being the culprit as the problem seems to occur even when no one is connected and for sure when the AGI is not active. It is the older AreskiCC version tho as I have had no incentive to upgrade it as we use it mainly for testing quality etc. Any clearer? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: queue - transfer calls
Hi Ariel, Thinking a little bit more about your idea of parking calls for 'simulating' a consultive transfer, I realized the following problem: If an agent is making an outgoing call (or even receiving a call that is not coming from the queue), he is not considered busy to the queue manager.] That means that once the agent parks a users call, if calls to his manager to tell him there is a parked call waiting to be answered, he immediately becomes available to the queue, and might receive calls even while he is talking to the manager. Is there a way to define that an agent is busy if he is on any call, not just calls coming from the queue? Thank you Dov Message: 9Date: Mon, 18 Apr 2005 10:18:31 -0400From: "Ariel Batista" [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] queue - transfer callsTo: "'Asterisk Users Mailing List - Non-Commercial Discussion'"asterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Content-Type: text/plain; charset="us-ascii"Hello,I am setting up an ACD using *, but found a an issue that I am not being able to resolve, and this might impact our * implementation.We have a call center with 4 agents, which should receive calls from theirqueue. But we also have a "call center management" team which should be able to talk to end customers in case the first level call center is not able tosolve the problem.There are two issues there:1. The agent cannot use the soft-phone TRANSFER button.. she has topress the pound key to transfer. This is not a 'terrible' issue, since it isjust a matter of educating agents.This one can be fixed if you want by going with the paid xten pro software.It has a transfer button.2. Attended transfer: If the agent transfers the call to someone in themanagement team, the call is immediately transferred, and the agent is notable to talk to the manager before. Is there a way to allow an agent to talkto the management befora actually transferring, so that he can explain theissue in advanceIn stead of transferring to the next level support have your agents park thecall to lets say 700 it should give you something like 701 then call thenext agent tell them what the problem is and to pickup exten 701.Thank you very muchDov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Unbelievable...
I have dealt with livevoip on several issues as the new account I just set up had a number of problems, unlike the first one I purchased. They were responsive, offered fixes within an hour, fixed their problems within the day, and I have had no problems or rude responses with them. I would tend to respect their wishes to have your problems sent directly rather than posted publicly, and only if you decide they are not working out would I post your grievances. Chris Mason www.anguillaguide.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] Snom subscribe/notify problem
I have a Snom-190 that I've successfully used on a * box with the LED's lighting up when a line goes active. I have moved it to another box, though, and I'm having trouble with it. It almost seems as though there is a limit to how long a sip channel name can be for the subscribe/notify to work right. If I have the following in sip.conf: -- [snom] type=friend ; Friends place calls and receive calls context=PewTest-snom ; Context for incoming calls from this user host=dynamic ; This peer register with us callerid=Snom190 201 dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info disallow=all allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! [EMAIL PROTECTED] ; Mailbox(-es) for message waiting indicator accountcode=PewTest amaflags=documentation; default AMA flag [PewTest-grandstream] type=friend ; either friend (peer+user), peer or user callgroup=1 ; We are in caller groups 1,3,4 pickupgroup=1 ; We can do call pick-p for call group 1,3,4,5 context=PewTest-internal username=grandstream1 ; usually matches the [section] title callerid=grandstream 202 host=dynamic ; we have a static but private IP address canreinvite=yes ; allow RTP voice traffic to bypass Asterisk dtmfmode=info ; either RFC2833 or INFO for the BudgeTone outgoinglimit=1 ; disable callwaiting signal (2nd call to phone) incominglimit=1 ; permit only 1 outgoing call at a time [EMAIL PROTECTED] disallow=all ; need to disallow=all before we can use allow= allow=ulaw; Note: In user sections the order of codecs accountcode=PewTest amaflags=documentation; default AMA flag -- and this in extensions.conf: -- [PewTest-snom] ;include = PewTest-internal ; extensions for monitoring exten = 200,hint,SIP/PewTest-sipura1 exten = 201,hint,SIP/snom exten = 202,hint,SIP/PewTest-grandstream exten = 203,hint,SIP/PewTest-grandstream exten = 200,1,Dial(SIP/PewTest-sipura1) exten = 201,1,Dial(SIP/snom) exten = 202,1,Dial(SIP/PewTest-grandstream) exten = 203,1,Dial(SIP/PewTest-grandstream) -- and the snom is set to light up it's LEDs for extensions 200-203. The LED's work just find when I call the snom (SIP/snom), but the light for the grandstream will not light up (SIP/PewTest-grandstream). If I change the entries for the grandstream from PewTest-grandstream to grandstream, then the light will work for that line, too. If I change the entries for the snom from snom to PewTest-snom, then the snom light fails to work. I have run sip debug mode on the snom peer and * is not sending out the NOTIFY messages, so it does not appear to be an issue with the Snom. Is there some type of limit to the SIP SUBSCRIBE/NOTIFY stuff that only allows 8 character channel names? Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] Calling Card
Anyone experimented with Calling Card support in * Am I wrong in presuming that if I have one calling card caller call in and want to complete a call I will use 2 lines (1 for the customers inbound and another to complete the remote call)?? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I use Asterisk for a modified Hoot and Holler?
Hello wonderful asterisk users list. I have some energy traders that are currently using 2 wire hoot-n-hollers (squawk box, always open direct line) to different trading floors throughout the country. Each box has one hoot-n-holler line. I would like to make these boxes IP based by connecting the 4 lines to a TDM400 with 4 fxo modules, providing SIP or IAX extensions for each hoot line and giving my clients a softclient on their desktop. Will it work? Is it reliable? Is it the best way? Thanks in advance for you Asterisk wisdom on this little problem of mine. Matthew Machen Southern Company Network Support ___ 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] Snom subscribe/notify problem
On Mon, Apr 18, 2005 at 11:34:03AM -0400, Michael George wrote: I have a Snom-190 that I've successfully used on a * box with the LED's lighting up when a line goes active. I have moved it to another box, though, and I'm having trouble with it. It almost seems as though there is a limit to how long a sip channel name can be for the subscribe/notify to work right. If I have the following in sip.conf: -- snip -- and this in extensions.conf: -- snip -- and the snom is set to light up it's LEDs for extensions 200-203. The LED's work just find when I call the snom (SIP/snom), but the light for the grandstream will not light up (SIP/PewTest-grandstream). If I change the entries for the grandstream from PewTest-grandstream to grandstream, then the light will work for that line, too. If I change the entries for the snom from snom to PewTest-snom, then the snom light fails to work. I have run sip debug mode on the snom peer and * is not sending out the NOTIFY messages, so it does not appear to be an issue with the Snom. Is there some type of limit to the SIP SUBSCRIBE/NOTIFY stuff that only allows 8 character channel names? It appears that the hyphen (-) in the channel name is what is breaking things. If I take that out, all seems to work fine. Anyone know why that might be? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] One-way audio
Hi all, Maybe someone encountered similar issue. I have an * with the incoming DID over SIP. * is behind a firewall. I have no issues with other SIP devices connected from the outside network, however on that DID when I receive a call I can hear only incoming audio, no outgoing. If I setup a playback with some audio stream, * just disconnects the call right after it receives it. The same issue happens no matter which client is being connected to that DID. For example: [inbound] exten = 225612,1,SetAccount(225612) exten = 225612,2,Ringing() exten = 225612,3,Dial(SIP/bt101,50) exten = 225612,4,Hangup If I change Dial(SIP/bt101,50) to Dial(IAX2/firefly,50) it does not change anything. This example can only receive audio. This one just answers and disconnects call the same second: [inbound] exten = 225612,1,SetAccount(225612) exten = 225612,2,Answer exten = 225612,3,Playback(vm-goodbye) exten = 225612,4,Hangup Sincerely, --Andy x6722 ___ 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] snom and hint priority
On Apr 17, 2005, at 12:23 AM, Lance Grover wrote: I have rebooted the phone and restarted asterisk after each change. Did you do it in that order? If so, that is probably a source of trouble (you should restart or reload asterisk before the phone boots, not after). -- Joshua P. Dady http://www.indecisive.com/ smime.p7s Description: S/MIME cryptographic signature ___ 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] Asterix Manager Proxy in Java/EJB?
Title: Asterix Manager Proxy in Java/EJB? Anyone doing/done a manager proxy to Asterisk in Java? Looking to avoid the Python/PERL/etc. managers (not that theres anything wrong with them or the languages) but were running a Java environment already and Id like to not re-invent the wheel if possible. Colin Stefani Tideworks Technology ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I use Asterisk for a modified Hoot and Holler?
What is it you're trying to accomplish? Squawk Box--fxo---*--IAX2/SIP clients? or replace the 2 wire solution between the different locations with IAX2/SIP? The only thing I'd caution you about hear is as you're going back and forth between 2 wire and IAX2/SIP along with conferencing/meetme in a hoot/holler application be real careful that you've worked all the kinks out with echo cancelation. Some echo thrown back into conference, as you probably already know, could get real ugly real quick. On 4/18/05, Machen, Matthew T. [EMAIL PROTECTED] wrote: Hello wonderful asterisk users list. I have some energy traders that are currently using 2 wire hoot-n-hollers (squawk box, always open direct line) to different trading floors throughout the country. Each box has one hoot-n-holler line. I would like to make these boxes IP based by connecting the 4 lines to a TDM400 with 4 fxo modules, providing SIP or IAX extensions for each hoot line and giving my clients a softclient on their desktop. Will it work? Is it reliable? Is it the best way? Thanks in advance for you Asterisk wisdom on this little problem of mine. Matthew Machen Southern Company Network Support ___ 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] Strange tones when placing a PSTN call.
I recently installed [EMAIL PROTECTED] and got one of the TDM400p cards configured to connect to my POTS line. I can make outgoing calls with no problem however I seem to have a short delay followed by 5 beeps before the line starts ringing out. Does anyone know what would cause this ? Michael Martin Systems Engineer Netranom Communications * email: [EMAIL PROTECTED] ( office: 304.562.4700 h mobile: 304.419.1510 : web: www.netranom.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] Lots of RTP checksum errors
Hi all, i'm getting about 1-3 NOTICE[1915]: rtp.c:453 ast_rtp_read: RTP: Received packet with bad UDP checksum message per call on CVS HEAD from 31 Mar. which seems some changes regarding rtpchecksums is made at that time. setting rtpchecksums to no or yes in rtp.conf doesn't make any sense. now i'm using latest CVS Head. any ideas? Thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940
Thomas: It sounds like you may need to unlock your phone. If I recall, you can hit **# to unlock it; then go to the settings menu. On newer firmwares, you'll have fun trying to get past an actual password. Check http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx as well as this list's archives from April 14,15,16,17 of this year. This topic just came up recently a few times. -Andy On 4/18/05, Thomas RULMONT [EMAIL PROTECTED] wrote: Hello list, Could you tell me if you ever succeeded in configuring Cisco 7940 and chan_skinny. How ? (I cannot configure my phone, almost any submenu is unavailable) Thx. -- Thomas RULMONT Responsable Commercial Alterys SA T. +32 87 325939 T. +32 486 863216 E. [EMAIL PROTECTED] ___ 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] Asterix Manager Proxy in Java/EJB?
Title: Asterix Manager Proxy in Java/EJB? Ok, I just answered my own question, for the edification of the group: http://www.voip-info.org/wiki-Asterisk-java Colin Stefani Tideworks Technology From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Stefani Sent: Monday, April 18, 2005 9:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterix Manager Proxy in Java/EJB? Anyone doing/done a manager proxy to Asterisk in Java? Looking to avoid the Python/PERL/etc. managers (not that theres anything wrong with them or the languages) but were running a Java environment already and Id like to not re-invent the wheel if possible. Colin Stefani Tideworks Technology ___ 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] Calling Card
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Huddleston, Robert Sent: Monday, April 18, 2005 10:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Calling Card Anyone experimented with Calling Card support in * Am I wrong in presuming that if I have one calling card caller call in and want to complete a call I will use 2 lines (1 for the customers inbound and another to complete the remote call)?? Thanks You are not wrong ___ 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] Can I use Asterisk for a modified Hoot and Holler?
Hello wonderful asterisk users list. I have some energy traders that are currently using 2 wire hoot-n-hollers (squawk box, always open direct line) to different trading floors throughout the country. Each box has one hoot-n-holler line. I would like to make these boxes IP based by connecting the 4 lines to a TDM400 with 4 fxo modules, providing SIP or IAX extensions for each hoot line and giving my clients a softclient on their desktop. Will it work? Is it reliable? Is it the best way? The probability of making that work with a TDM card is rather low. The primary reason is the TDM card (and drivers) are oriented around ringing come in, some action performed (eg, dialplan), and aswering the call. Hoot-n-holler circuits don't have those same functions. In most cases that I'm familiar with, the actual circuit used for these is four-wire (not 2-wire) and the TDM does not have a four-wire interface. Could someone modify the drivers to do that? High probability, but that isn't going to be an easy task for the uninitiated. ___ 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] Calling Card
Huddleston, Robert wrote: Anyone experimented with Calling Card support in * Am I wrong in presuming that if I have one calling card caller call in and want to complete a call I will use 2 lines (1 for the customers inbound and another to complete the remote call)?? It depends wether you are using POTS or VOIP to terminate the call. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IPswitch: How to use speed dialing?
Hi Ronald,
It seems like you need to put in default as your context. However I think
your problem was that you put the number in CallerID column and The CallerID
in the Name column. I was hoping to hear if it helped you to change that?
Thorben
Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse
news:[EMAIL PROTECTED]
tgj wrote:
Hi Ronald,
I must admit I am getting confused now.
I understand that you have a problem getting Speed Dial Buttons to work.
The problem as I understand it is that the calls are placed in the wrong
context.
To solve that problem I have asked you to make sure that you have typed a
valid context on the configuration page. Have you tried that?
I think thats all you need to do, how do I post an example of that? It's a
fairly easy thing to do.
Thorben
What is the right syntax to do that?
Context for dialing a trunk line is trunkint
Peter has the phone number 011-234-5678
How to set it up as a speed dial number? Below are all info you may need:
The phone 601 (= Monitor extension) is a Sip phone,
[general]
context=default; Default context for incoming calls
[601]
type=friend
username=601
secret=dont+tell+you
canreinvite=no
host=dynamic
dtmfmode=rfc2833
[EMAIL PROTECTED]
nat=yes
callgroup=1
pickupgroup=1
callerid=Ronald Hotline,601
qualify=1000
extensions.conf
[default]
...
include = trunkint
...
[trunkint]
;
; International long distance through trunk
; . other lines deleted
exten = _9011Z.,107,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9011Z.,108,hangup
bye
Ronald
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Re: [Asterisk-Users] Help compiling zaptel in Debian
With auto-apt the problem is not solved... Thanks Andres Paglayan escribió: try auto-apt for getting dependencies satisfied on the fly while compiling. Manuel Casal wrote: During the zaptel configuration at the end of it there is this error: ___ 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] Calling Card
Yes, if I understand what you are asking. 1. The Card User calls to your asterisk PBX. 2. Asterisk answers the call on line 1. 3. Asterisk places an outgoing call on line 2 bridging the lines. (That is how it works in the SIP world.) So you would need an FXO/FSO pair of lines to let them make a call in the analog/digital/TDM world. Now, as to the definition of a line: Line 1 is a PSTN line because they are using a calling card to get to your Asterisk PBX. Line 2 could be a VoIP line to another Asterisk PBX or like PBX/switch. In this case you only need 1 PSTN/Analog/DS0 type line to receive incoming calls. --- If you provide the calling card person with a VoIP phone, then you don't need the PSTN lines. Because both line 1 and line 2 are IP connections. My guess is that you want to start a calling card business based on VoIP. Then you need PSTN (Telephone Company) lines to collect the incoming PSTN calls and convert those calls to the internet VoIP calls. If you want to host termination of calls, that is where calls come long distance into your PBX and you convert them back into local calls. This is where you partner with another calling card provider and he terminates his calls to your phone lines locally. The two of you trade off call termination; the one who makes the most calls pays the difference to the other guy for using his lines for out going calls to the PSTN lines. Is that your questions? Either way you need to remember to budget for a billing system. Race The Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Huddleston, Robert Sent: Monday, April 18, 2005 11:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Calling Card Anyone experimented with Calling Card support in * Am I wrong in presuming that if I have one calling card caller call in and want to complete a call I will use 2 lines (1 for the customers inbound and another to complete the remote call)?? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to specify channel 1: No such device
Hi, I did not find any useful information to configure a Wildcard
TDM400P with a FXO card. I've tried everithing, I tried configure it
using the cvs and the information from digium page, I tried to
configure it using
debian packages, I tried to configure with kernels 2.4.30 and 2.6.11, I even
switched the mother board (I tried 3 motherboards).
I tanks in advace any help you could give me.
Best Regards,
Gregorio Toscano
[EMAIL PROTECTED]
The erros are:
Apr 15 16:08:37 WARNING[1468]: chan_zap.c:850 zt_open: Unable to
specify channel 1: No such device
Apr 15 16:08:37 ERROR[1468]: chan_zap.c:6458 mkintf: Unable to open
channel 1: No such device
here = 0, tmp-channel = 1, channel = 1
Apr 15 16:08:37 ERROR[1468]: chan_zap.c:9558 setup_zap: Unable to
register channel '1'
My configuration files are:
lsmod
Module Size Used byNot tainted
wctdm 97248 0 (unused)
zaptel214784 0 [wctdm]
dmesg (final):
Module 0: Not installed
Module 1: Not installed
Module 2: Not installed
Module 3: Installed -- AUTO FXO (FCC mode)
Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)
najay:/etc# cat zaptel.conf
loadzone = us
defaultzone = us
fxs_ks=1
najay:/etc/asterisk# cat zapata.conf
[channels]
signalling=fxo_ks
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call,
number is in milliseconds
callerid=asreceived
group=1
context=default ; Points to the default context of your extensions.conf
channel = 1 ; Again X is the number of FXO modules you have
najay:/etc/asterisk# cat voicemail.conf
[general]
format=wav
[default]
8500 = 1234,Gummer,[EMAIL PROTECTED]
[root at fred asterisk]#cat extensions.conf
[default]
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
exten = 0.,1,Dial(Zap/g1/${EXTEN}/20,t)
exten = s,1,Wait(1)
exten = s,2,Dial,Zap/g1 ; Dials the first available channel in group 1
exten = s,3,Voicemail,u9000
exten = s,4,Hangup
najay:/proc# cat interrupts
CPU0
0: 362024 XT-PIC timer
1: 6987 XT-PIC keyboard
2: 0 XT-PIC cascade
10:2792513 XT-PIC wctdm
11: 10280 XT-PIC via82cxxx, eth0
12: 3653 XT-PIC PS/2 Mouse
14: 16860 XT-PIC ide0
NMI: 0
ERR: 0
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RE: [Asterisk-Users] Unable to specify channel 1: No such device
Where is this line in zapata.conf under the [channels] context?
channel=1
Also is that line in zaptel.conf correct? Here is mine Note the
lack of and underscore on fxsks...
fxsks=1
loadzone = us
defaultzone=us
Try these settings and the run ztcfg -vvv
Restart * and see what you get then
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregorio
Toscano
Sent: Monday, April 18, 2005 9:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Unable to specify channel 1: No such device
Hi, I did not find any useful information to configure a Wildcard
TDM400P with a FXO card. I've tried everithing, I tried configure it
using the cvs and the information from digium page, I tried to configure
it using debian packages, I tried to configure with kernels 2.4.30 and
2.6.11, I even switched the mother board (I tried 3 motherboards).
I tanks in advace any help you could give me.
Best Regards,
Gregorio Toscano
[EMAIL PROTECTED]
The erros are:
Apr 15 16:08:37 WARNING[1468]: chan_zap.c:850 zt_open: Unable to specify
channel 1: No such device Apr 15 16:08:37 ERROR[1468]: chan_zap.c:6458
mkintf: Unable to open channel 1: No such device here = 0, tmp-channel
= 1, channel = 1 Apr 15 16:08:37 ERROR[1468]: chan_zap.c:9558 setup_zap:
Unable to register channel '1'
My configuration files are:
lsmod
Module Size Used byNot tainted
wctdm 97248 0 (unused)
zaptel214784 0 [wctdm]
dmesg (final):
Module 0: Not installed
Module 1: Not installed
Module 2: Not installed
Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM:
Wildcard TDM400P REV E/F (4 modules)
najay:/etc# cat zaptel.conf
loadzone = us
defaultzone = us
fxs_ks=1
najay:/etc/asterisk# cat zapata.conf
[channels]
signalling=fxo_ks
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your
needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number
is in milliseconds callerid=asreceived
group=1
context=default ; Points to the default context of your extensions.conf
channel = 1 ; Again X is the number of FXO modules you have
najay:/etc/asterisk# cat voicemail.conf
[general]
format=wav
[default]
8500 = 1234,Gummer,[EMAIL PROTECTED]
[root at fred asterisk]#cat extensions.conf
[default]
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
exten = 0.,1,Dial(Zap/g1/${EXTEN}/20,t) exten = s,1,Wait(1) exten =
s,2,Dial,Zap/g1 ; Dials the first available channel in group 1 exten =
s,3,Voicemail,u9000 exten = s,4,Hangup
najay:/proc# cat interrupts
CPU0
0: 362024 XT-PIC timer
1: 6987 XT-PIC keyboard
2: 0 XT-PIC cascade
10:2792513 XT-PIC wctdm
11: 10280 XT-PIC via82cxxx, eth0
12: 3653 XT-PIC PS/2 Mouse
14: 16860 XT-PIC ide0
NMI: 0
ERR: 0
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Re: [Asterisk-Users] 99% CPU - CVS 03.28.05
I've been running a version of the CVS without issue until late last week when suddenly Asterisk would randomly hit 99% CPU and stop registering my DIDs. Similar things happened to me with the CVS version from around that time. Randomly every 2-3 days asterisk would use 99% CPU and just sit there. Usually happened during night, though, when no calls were active. No errors, no warnings. Eventually the console said Registration timed out a few time, but then it was completely dead. I had to kill asterisk with the -9 signal. My suspicion is that there is a bug in chan_sip that was introduced when the jitterbuffer stuff was added. Going back to a CVS version from mid-Feb works perfectly fine for me, without any other changes. --Luki ___ 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] Still having broadvoice issues
disallow=all allow=g726 allow=g729 Change to this and try again: disallow=all allow=ulaw Broadvoice officially only supports ulaw. g726 works some times on some numbers, but don't rely on that. You can also drop the callerid= since Broadvoice will not use it anyway. --Luki ___ 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] Hitachi WIP-5000/IP-5000 firmware
Good suggestion. It now seems to roam between access points nicely, even while a call is in progress. What access pooints are you using? -rb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco/Asterisk codec negotiation problems
As a followup for any who has the same problem, and searches the archives (don't you love finding the problem you have in the archive, but no-one followed it up?), check the following references: http://lists.digium.com/pipermail/asterisk-dev/2005-April/011291.html and the status of the updated code: http://bugs.digium.com/bug_view_page.php?bug_id=0003346 Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ Alistair Cunningham wrote: On more testing, I conclude that Asterisk isn't being very clever about codec negotiation. My understanding (possibly faulty) from experiments is this. If I have: UA1 -- Asterisk -- UA2 and have disallow/allow entries in UA1's stanza in sip.conf, it seems that the first entry in the allow list is all that's used to choose the codec from UA1. Entries in UA2's stanza and SIP responses from UA2 are not used. If it turns out that UA2 can't support the codec that Asterisk chose for UA1, Asterisk attempts a translation. This occurs even if UA1 and UA2 have a supported codec in common which isn't the one Asterisk chose. If my understanding is correct, this is very inefficient. Worse, if one of the codecs is one it doesn't understand, such as G.729 (without chan_g729a.so) or G.723.1, Asterisk drops the call, even though it could have done pass through. Is my understanding correct? Is this a weakness in Asterisk? Am I missing something elementary? ___ 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] Help compiling zaptel in Debian
Manuel, This is from my Wiki page on running Asterisk on Debian/GNU Linux. Build and Install Zaptel Zaptel provides support for Digium hardware. The following steps can be followed to build and install Zaptel. 1. Create symbolic links to the new kernel's source files by issuing the following commands at a console window's command line: cd /usr/src ln -s /usr/src/kernel-source-2.4.20 linux ln -s /usr/src/kernel-source-2.4.20 linux-2.4 2. Build and install Zaptel by issuing the following commands at a console window's command line: cd /usr/src/zaptel make clean; make install I'm pretty sure that will solve your problem. My Wiki page has been stagnant for a while, because I've been dealing with the rather steep Asterisk learning curve, but I've got a ton of notes on paper that I'll be adding to the page once I hit my next documentation stage. Matthew Roth http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian Manuel Casal wrote: ...snip... depmod: *** Unresolved symbols in /lib/modules/2.4.20-686/misc/ztd-eth.o [ `id -u` = 0 ] /sbin/depmod -a || : depmod: *** Unresolved symbols in /lib/modules/2.4.20-686/misc/ztd-eth.o [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample /etc/zaptel.conf I have previously made the make oldconfig and the make dep ... I'm using debian with 2.4.20 kernel... any ideas..? thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to specify channel 1: No such device
Inline...
Hi, I did not find any useful information to configure a Wildcard
TDM400P with a FXO card. I've tried everithing, I tried configure it
using the cvs and the information from digium page, I tried to
configure it using
debian packages, I tried to configure with kernels 2.4.30 and 2.6.11, I even
switched the mother board (I tried 3 motherboards).
I tanks in advace any help you could give me.
Best Regards,
Gregorio Toscano
[EMAIL PROTECTED]
The erros are:
Apr 15 16:08:37 WARNING[1468]: chan_zap.c:850 zt_open: Unable to
specify channel 1: No such device
Apr 15 16:08:37 ERROR[1468]: chan_zap.c:6458 mkintf: Unable to open
channel 1: No such device
here = 0, tmp-channel = 1, channel = 1
Apr 15 16:08:37 ERROR[1468]: chan_zap.c:9558 setup_zap: Unable to
register channel '1'
My configuration files are:
lsmod
Module Size Used byNot tainted
wctdm 97248 0 (unused)
zaptel214784 0 [wctdm]
dmesg (final):
Module 0: Not installed
Module 1: Not installed
Module 2: Not installed
Module 3: Installed -- AUTO FXO (FCC mode)
Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)
najay:/etc# cat zaptel.conf
loadzone = us
defaultzone = us
fxs_ks=1
^^ that should be fxsks (might also try fxsks=3 since your only
module is #3. I don't remember how these are numbered for sure.
Don't forget to run 'ztcfg -vv' after the modprobes. That should
tell you which channel the fxo module is on.
najay:/etc/asterisk# cat zapata.conf
[channels]
signalling=fxo_ks
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call,
number is in milliseconds
callerid=asreceived
group=1
context=default ; Points to the default context of your extensions.conf
^^^ I strongly suggest changing this to some other keyword
as it appears that you're using default as the sip
context as well, and that's going to give you fits.
Try something like context=inbound-home or whatever.
channel = 1 ; Again X is the number of FXO modules you have
najay:/etc/asterisk# cat voicemail.conf
[general]
format=wav
I assume the following is really extensions.conf (even though you didn't
mention it).
[default]
8500 = 1234,Gummer,[EMAIL PROTECTED]
[root at fred asterisk]#cat extensions.conf
[default]
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
exten = 0.,1,Dial(Zap/g1/${EXTEN}/20,t)
exten = s,1,Wait(1)
exten = s,2,Dial,Zap/g1 ; Dials the first available channel in group 1
exten = s,3,Voicemail,u9000
exten = s,4,Hangup
Now you need to add this for inbound TDM calls (see above):
[inbound-home]
exten = s,1,Dial(SIP/8500},15) ; extn to ring when inbound call arrives.
exten = s,103,Congestion
najay:/proc# cat interrupts
CPU0
0: 362024 XT-PIC timer
1: 6987 XT-PIC keyboard
2: 0 XT-PIC cascade
10:2792513 XT-PIC wctdm
11: 10280 XT-PIC via82cxxx, eth0
12: 3653 XT-PIC PS/2 Mouse
14: 16860 XT-PIC ide0
NMI: 0
ERR: 0
Interrupts look fine with wctdm card on its own.
Rich
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[Asterisk-Users] Only one PRI out of four working on TE405p?
Hi everyone, I'm struggling to get four E1 primary rate ISDN lines working in a * server with a TE405p. So far almost so good... My configuration files are below but my problem seems to be that only 30 B-channels are being seen by asterisk - when I start * with -vvvgc I get the following as the last debug items: - -- B-channel 0/1 successfully restarted on span 1 -- B-channel 0/2 successfully restarted on span 1 ... ... -- B-channel 0/31 successfully restarted on span 1 (30 lines in total of course) But I'm not seeing the other 90 channels identified and I'm not sure what's wrong in my configuration? Any help much appreciated! Thanks, Derek My /etc/zaptel.conf : span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124 dchan=16 My /etc/asterisk/zapata.conf: [trunkgroups] trunkgroup = 1,16 spanmap = 1,1,0 spanmap = 2,1,1 spanmap = 3,1,2 spanmap = 4,1,4 [channels] context=PRI-NTL switchtype=euroisdn signalling=pri_cpe group=1 usecallerid=yes hidecallerid=no callwaiting=no restrictcid=no threewaycalling=no immediate=yes channel=1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124 -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.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] Voicemail not working...
Title: Voicemail not working... Hello All, My voicemail seems to have stopped working and I cannot figure out why. After call times out, the user receives a message the no one is available to take the call. The CLI shows this... -- Got SIP response 603 Decline back from 192.168.1.248 Then the user is disconnected. My VM was working fine and nothing has been changed. Anyone know what could cause this? I am on AAH 0.6 Thanks, Wiley ___ 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] TDM400P Revision question.
-Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] As far as the issue with DC voltage on the POTS line only being 43.8 DC, my guess was that is just an issue with voltage drop on the line because of distance between me and the CO. No possible way. If everything is truly on hook, there isn't any current draw and therefore no way for a voltage drop to occur. Basic ohm's law. If he's not using a high-impedance voltmeter, the meter might be loading the circuit down. ___ 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] 99% CPU - CVS 03.28.05
No changes were made to chan_sip when the iax2 jitter buffer was added. However, ive seen and hear several complaints about coredumps, deadlocks in cvs-head chan_sip recently. /Z Luki wrote: I've been running a version of the CVS without issue until late last week when suddenly Asterisk would randomly hit 99% CPU and stop registering my DIDs. Similar things happened to me with the CVS version from around that time. Randomly every 2-3 days asterisk would use 99% CPU and just sit there. Usually happened during night, though, when no calls were active. No errors, no warnings. Eventually the console said Registration timed out a few time, but then it was completely dead. I had to kill asterisk with the -9 signal. My suspicion is that there is a bug in chan_sip that was introduced when the jitterbuffer stuff was added. Going back to a CVS version from mid-Feb works perfectly fine for me, without any other changes. --Luki ___ 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 signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: IPswitch: How to use speed dialing?
tgj wrote:
Hi Ronald,
It seems like you need to put in default as your context. However I think
your problem was that you put the number in CallerID column and The CallerID
in the Name column. I was hoping to hear if it helped you to change that?
Let's try it together:
1. Open IPswitch
2. Open Extensions tab on top
3. Switch to the tab Speed Dials on the bottom
4. Fill in:
Name: [EMAIL PROTECTED]
Caller Id: Peter
Visible on Panel: (ticket)
Exentension Group: Speed Dial Numbers
Congratualtions, you have successfully installed the Asterisk Open
Source .
bye
Ronald
Thorben
Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse
news:[EMAIL PROTECTED]
tgj wrote:
Hi Ronald,
I must admit I am getting confused now.
I understand that you have a problem getting Speed Dial Buttons to work.
The problem as I understand it is that the calls are placed in the wrong
context.
To solve that problem I have asked you to make sure that you have typed a
valid context on the configuration page. Have you tried that?
I think thats all you need to do, how do I post an example of that? It's a
fairly easy thing to do.
Thorben
What is the right syntax to do that?
Context for dialing a trunk line is trunkint
Peter has the phone number 011-234-5678
How to set it up as a speed dial number? Below are all info you may need:
The phone 601 (= Monitor extension) is a Sip phone,
[general]
context=default; Default context for incoming calls
[601]
type=friend
username=601
secret=dont+tell+you
canreinvite=no
host=dynamic
dtmfmode=rfc2833
[EMAIL PROTECTED]
nat=yes
callgroup=1
pickupgroup=1
callerid=Ronald Hotline,601
qualify=1000
extensions.conf
[default]
...
include = trunkint
...
[trunkint]
;
; International long distance through trunk
; . other lines deleted
exten = _9011Z.,107,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9011Z.,108,hangup
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Re: [Asterisk-Users] Only one PRI out of four working on TE405p?
Thanks, Derek My /etc/zaptel.conf : span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124 dchan=16 My /etc/asterisk/zapata.conf: [trunkgroups] trunkgroup = 1,16 spanmap = 1,1,0 spanmap = 2,1,1 spanmap = 3,1,2 spanmap = 4,1,4 My guess is your not using NFAS so you need to forget about trunkgroups and just define the 4 E1s individually. [channels] context=PRI-NTL switchtype=euroisdn signalling=pri_cpe group=1 usecallerid=yes hidecallerid=no callwaiting=no restrictcid=no threewaycalling=no immediate=yes channel=1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124 -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Can I use Asterisk for a modified Hoot and Holler?
Hello wonderful asterisk users list. I have some energy traders that are currently using 2 wire hoot-n-hollers (squawk box, always open direct line) to different trading floors throughout the country. Each box has one hoot-n-holler line. I would like to make these boxes IP based by connecting the 4 lines to a TDM400 with 4 fxo modules, providing SIP or IAX extensions for each hoot line and giving my clients a softclient on their desktop. Will it work? Is it reliable? Is it the best way? The probability of making that work with a TDM card is rather low. The primary reason is the TDM card (and drivers) are oriented around ringing come in, some action performed (eg, dialplan), and aswering the call. Hoot-n-holler circuits don't have those same functions. In most cases that I'm familiar with, the actual circuit used for these is four-wire (not 2-wire) and the TDM does not have a four-wire interface. Could someone modify the drivers to do that? High probability, but that isn't going to be an easy task for the uninitiated. Asterisk can make this happen, but you might have to ditch your current equipment. I'm assuming, though, that the reason you want to do this is to eliminate the costs of the lines. I think you'll really need to justify the cost of using bandwidth instead, buying the asterisk boxes, setting up QoS, time to configure. Asterisk is not a trivial thing to set up, and I would only do it if it's going to do something more than just this one function. The idea behind asterisk is that it does many things any connects many things. We do something similar with Polycom SIP phones, but it is not always on. PSTN phone networks have gotten to the point where they are (supposedly) reliable to five nines (99.999%). IP networks are NOT there, so you'd probably have to do things to make sure that your IP hoot-n-holler setup is self-healing in case your connections do ever go down. If you use your IP connections for other traffic, there will also be QoS issues to deal with. You'll probably have to make sure your voice traffic is getting priority over everything else, or there will likely be audio quality issues. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can I use Asterisk for a modified Hoot andHoller?
Hmmm, Hoot and Holler, hoot-n-holler, ARD: Automatic Ring Down, Hot line and Private Line Automated Ringdown (PLAR) You should think about VoIP via Asterisk. Here is a quick search result on it http://lists.digium.com/pipermail/asterisk-users/2003-March/008936.html But not much else. I am looking at pre-dial on the cisco 7960 or the snom 200's I have. This might be a good thing to add to sip. Just need to get the phone to ping Asterisk when it goes off hook. A few changes to the sip C code to make it dial an extension/context. If someone knows how to get a voip phone to pre-dail then the SIP changes are not that hard. Race The Tyrant Vanderdecken Looking at this issue I stole/lifted this from a hoot-n-holler company. An Auto-Ring Down is a leased voice circuit that connects two single endpoints together. When either telephone handset is taken off-hook, the remote telephone automatically rings. This application is used most frequently for brokerage firms, banks, Wall Street firms and applications that require immediate verbal responses. Since ARDs and Hoot'n'Hollers are popular with Wall Street and Recycling Companies, (name of company removed here) has specialized in both types of circuits. This type of voice-grade analog circuit is considered a specialty circuit in that the Bell codes for signaling on both ends are not coded the same on each side, as are most other types of circuits. This is called a 2-state signaling scheme, based upon the use of the A-Bit. Like many other voice-grade analog circuits, it can be ordered either as a 2 or 4 wire connection based on the requirement of the customer's terminating equipment. If any of the segments of the tail circuits into the customer's premise are in excess of 6 miles, the circuit should be ordered up as a 4 wire. This is followed with a hybrid station pack at the customer site to break it back down to 2 wires for their equipment. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, April 18, 2005 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can I use Asterisk for a modified Hoot andHoller? Hello wonderful asterisk users list. I have some energy traders that are currently using 2 wire hoot-n-hollers (squawk box, always open direct line) to different trading floors throughout the country. Each box has one hoot-n-holler line. I would like to make these boxes IP based by connecting the 4 lines to a TDM400 with 4 fxo modules, providing SIP or IAX extensions for each hoot line and giving my clients a softclient on their desktop. Will it work? Is it reliable? Is it the best way? The probability of making that work with a TDM card is rather low. The primary reason is the TDM card (and drivers) are oriented around ringing come in, some action performed (eg, dialplan), and aswering the call. Hoot-n-holler circuits don't have those same functions. In most cases that I'm familiar with, the actual circuit used for these is four-wire (not 2-wire) and the TDM does not have a four-wire interface. Could someone modify the drivers to do that? High probability, but that isn't going to be an easy task for the uninitiated. ___ 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] Calling Card
Anyone experimented with Calling Card support in * Am I wrong in presuming that if I have one calling card caller call in and want to complete a call I will use 2 lines (1 for the customers inbound and another to complete the remote call)?? If you use IAX2 termination for incoming and outgoing calls (Voicepulse for instance), the call, once established will be natively bridged on your providers network, not using any resources on your own * box. ___ 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
