Re: [asterisk-users] Finding difficulty in installing Asterisk
On Friday 25 January 2008 05:25:57 Lyle Giese wrote: You need to do a 'make' before the 'make install'. make install will do all that is necessary to install a program including making any files necessary. -- Dave Cotton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: dtmf mode
Hi Jarga, What type of connection you are using is it VoIP or ISDN PRI, if it is VoIP check your dtmfmode in sip.conf if it is PRI check zapata.conf On Jan 25, 2008 12:13 AM, Jarga Jallow [EMAIL PROTECTED] wrote: Hi, I am having trouble making a selection when I call a number and need to make a selection to go to an extension with my polycom phones 301. Anybody have an idea how to fix this problem? Thanks in advance. Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 http://www.2mcctv.com/ www.2mcctv.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in image003.gifimage002.gifimage001.jpg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding difficulty in installing Asterisk
Hi Pandey, What type of OS you are using, is it redhat or fedora. and install with latest version. On Jan 25, 2008 9:55 AM, Lyle Giese [EMAIL PROTECTED] wrote: You need to do a 'make' before the 'make install'. Lyle [EMAIL PROTECTED] wrote: Hi all, Please help me in installing Asterisk. I am getting the following error when trying to install Libpri [EMAIL PROTECTED] Asterisk]$ cd libpri-1.4.2 [EMAIL PROTECTED] libpri-1.4.2]$ make clean rm -f *.o *.so *.lo *.so.1 *.so.1.0 rm -f testprilib libpri.a libpri.so.1.0 rm -f pritest pridump rm -f .depend [EMAIL PROTECTED] libpri-1.4.2]$ make install gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -c -o copy_string.o copy_string.c ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding difficulty in installing Asterisk
Hi, I am using Fedora and I tried to install Libpri-1.4.3 but now I am getting error AVC access denied. Its saying I need to disable SELinux protection. I do not know what to do. Please help me out. Thanking you, Preeta -Original Message- From: [EMAIL PROTECTED] on behalf of Gopal krishnan Sent: Fri 1/25/2008 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Finding difficulty in installing Asterisk Hi Pandey, What type of OS you are using, is it redhat or fedora. and install with latest version. On Jan 25, 2008 9:55 AM, Lyle Giese [EMAIL PROTECTED] wrote: You need to do a 'make' before the 'make install'. Lyle [EMAIL PROTECTED] wrote: Hi all, Please help me in installing Asterisk. I am getting the following error when trying to install Libpri [EMAIL PROTECTED] Asterisk]$ cd libpri-1.4.2 [EMAIL PROTECTED] libpri-1.4.2]$ make clean rm -f *.o *.so *.lo *.so.1 *.so.1.0 rm -f testprilib libpri.a libpri.so.1.0 rm -f pritest pridump rm -f .depend [EMAIL PROTECTED] libpri-1.4.2]$ make install gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -c -o copy_string.o copy_string.c ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Maximum Paging Group Size?
Has anyone experience with (or an educated guess of) the largest paging group that can be supported by the Page() command? We have an installation coming up with 110 phones -- any hope to page this entire facility? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding difficulty in installing Asterisk
Hi Dave, I did make clean and then make. But then when I am giving make install its giving error AVC access denied. I am using Fedora. What may be the problem? Help me.. Thanking you, Preeta Pandey -Original Message- From: [EMAIL PROTECTED] on behalf of Dave Cotton Sent: Fri 1/25/2008 1:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Finding difficulty in installing Asterisk On Friday 25 January 2008 05:25:57 Lyle Giese wrote: You need to do a 'make' before the 'make install'. make install will do all that is necessary to install a program including making any files necessary. -- Dave Cotton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Share accounts several AOR
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Good morning, Is it possible with asterisk to allow to share the same account on 2 different devices, for example I want both my fix phone and my wifi phone to ring in the same time. I want to do it without making ringroups... Any idea how to do it? Thanks -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHma77rxOjjFYWQtoRAv6+AKCXqImQPJK0NxXHZlJDu6BShelwJwCeKVtj AAPzlXluS9e3t1qPXqA6sPU= =fpsa -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need sample configuration files for sipura/linksys ata
Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What kind of configuration do I need to run Asterisk ?
Hi, I hope someone in the mailing list has a good experience in server's configuration requirement since I was not able to make a large scale test to know Asterisk's configuration requirements for my application. So, I'd like to know what kind of configuration the following application should require : Asterisk'll be used as a VoIP server to handle video and audio communications between two computers connected to the Internet. The codecs used for video and audio are H263 and G711. The protocols used are RTP and SIP. Finally, Asterisk should be able to handle between 5 to 10 communications simultaneously. So do you know what kind of processor I should use ? How much memory ? And how much bandwidth ? Thank you, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Billing
Hello, I'm checking some Billing Software for Asterisk. In opensource I only know (the name, I haven't used) AstBill. What other software should I check with similar capabilities? Thank you! -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error in sip channel when asterisk created call (SIP invite request) is forked
Hi,
I encountered the following problem:
My asterisk works as a gateway between two sip networks external (public
internet) and internal (local lan)
From public side asterisk is registered UA in external network.
Internal sip UAs are registered in local SIP Proxy.
When there is an incoming call from external network some extension in
dialplan is invoked.
After this asterisk verifies mapping between dialed external user (invoked
extension) and target user in internal network
it dials internal user with the usage of following marco:
exten = s,1,NoOp(-- Internal call to: ${MACRO_EXTEN})
exten = s,n,Answer()
exten = s,n,Dial(SIP/${ARG1},60,ftT)
exten = s,n(congestion),Congestion(20)
This causes that call is redirected to the local sip proxy.
Error arises when invite request is forked in the sip proxy.
Due to the forked request there is an ringing response from
two user agents and those responses create early dialog
(they contain tag in from header field).
Ringing responses of course are received in some order to asterisk.
If call is answered by the user agent whose SIP ringing response came as
the second one asterisk never replies with the ACK.
If call is answered by the user agent whose SIP ringing response came as the
first
one everything works fine.
This is an error.
According to RFC 3261 Ringing response creates early dialog and if there is
no final response received
dialog should be terminated.
Is there any patch for this issue or a way to fix this through
configuration?
Bests regards
TOmasz
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Re: [asterisk-users] Asterisk Billing
Carles, You can find more info about the Open Source billing alternatives in the voip-info wiki: http://www.voip-info.org/wiki/view/Asterisk+billing Regards, Ariel On Fri, 2008-01-25 at 12:26 +0100, Carles Pina i Estany wrote: Hello, I'm checking some Billing Software for Asterisk. In opensource I only know (the name, I haven't used) AstBill. What other software should I check with similar capabilities? Thank you! -- Ariel Monaco Tel.: +49 (0) 2161 / 4643 - 0 credativ GmbH, HRB Mönchengladbach 12080 Hohenzollernstr. 133, 41061 Mönchengladbach Geschäftsführung: Dr. Michael Meskes, Jörg Folz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Billing
2008/1/25, Carles Pina i Estany [EMAIL PROTECTED]: Hello, I'm checking some Billing Software for Asterisk. In opensource I only know (the name, I haven't used) AstBill. What other software should I check with similar capabilities? Thank you! -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.kolmisoft.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unprovisioned 7961
Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ³Error Verifying Config Info². I have read quite a bit on this topic (getting 7961¹s to work with Asterisk and TB) and only came across a few postings where other people encountered this issue but no solution was given. I have checked the SEP.cnf.xml file for the phone and everything seems to be right. I even tried to remove some parts of the code as people suggested but no luck. I already have a 7960 on TB so I know that TFTP is working correctly. Any ideas on how I can get this to work would be much appreciated. Thank. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Adaptive jitterbuffer problem
Hello there, I have set a simple environment to test some functionalities of asterisk's new jitterbuffer. The environment is composed of a sip softphone registering in asterisk 1.4 and calling a pstn phone connected to asterisk through a fxs board. Using the fixed buffer implementation the call quality is improved when injecting a artificial jitter in my local network. However, when changing to the adaptive buffer and making the same call in the same environment, no audio is received in my phone and I get a warning in asterisk console: abstract_jb.c: Failed to put first frame in the jitterbuffer on channel ZAP. This is my zapata.conf: [trunkgroups] [channels] context=from-pbx signalling=fxo_ks usecallerid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 busydetect=yes busycount=6 channel=1 jbenable=yes jbimpl=adaptive Has anyone experienced such things? Any tips? Thanks in advance, folks. -- []'s André de Abrantes ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with FollowMe
BJ,
Yeah, that's what I figured from the code. But I still can't get my
hard coded #'s to work.
The line:
number = 201XXX,5
(201XXX is a US based phone # I want dialed that I've obscured
with X's - in case that wasn't originally clear)
This line reacts the same way as the AstDB configured lines.
Thanks,
Mike
On Jan 25, 2008, at 1:55 PM, BJ Weschke wrote:
Mike Coakley wrote:
I'm trying to use the FollowMe app with Asterisk 1.4.17. I've
followed
the WIKI page on setting it up but I can't seem to get it to work.
Here is my Asterisk version:
pbx1*CLI core show version
Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on
2008-01-10
12:08:48 UTC
Here is a log of when the FollowMe is being called:
NOTE: I've tried to use the AstDB as the WIKI described but when I
looked at the code it didn't seem like that would work so I've hard
coded a phone # for now. But I would prefer to use the AstDB if that
is a workable solution.
-- Executing [EMAIL PROTECTED]:8] FollowMe(Zap/
1-1,2000|a) in new stack
-- Zap/1-1 Playing 'vm-rec-name' (language 'en')
-- Zap/1-1 Playing 'beep' (language 'en')
-- x=0, open writing: /var/spool/asterisk/followme.1201278830.10
format: sln, 0x81c0218
-- User ended message by pressing #
-- Zap/1-1 Playing 'auth-thankyou' (language 'en')
-- Zap/1-1 Playing 'followme/pls-hold-while-try' (language 'en')
-- Started music on hold, class 'default', on channel 'Zap/1-1'
[Jan 25 11:34:32] NOTICE[23725]: chan_local.c:571 local_alloc: No
such
extension/context [EMAIL PROTECTED] creating local channel
[Jan 25 11:34:32] WARNING[23725]: app_followme.c:863 findmeexec:
Unable to allocate a channel for Local/FM1/[EMAIL PROTECTED] cause:
Unknown
[Jan 25 11:34:32] NOTICE[23725]: chan_local.c:571 local_alloc: No
such
extension/context [EMAIL PROTECTED] creating local channel
[Jan 25 11:34:32] WARNING[23725]: app_followme.c:863 findmeexec:
Unable to allocate a channel for Local/FM2/[EMAIL PROTECTED] cause:
Unknown
[Jan 25 11:34:32] NOTICE[23725]: chan_local.c:571 local_alloc: No
such
extension/context [EMAIL PROTECTED] creating local channel
[Jan 25 11:34:32] WARNING[23725]: app_followme.c:863 findmeexec:
Unable to allocate a channel for Local/FM3/[EMAIL PROTECTED] cause:
Unknown
(NOTE: This log was taken just prior to making the changes indicated
below in my Followme.conf file. But the changes reported the same
logs
lines, simply with different values (i.e. same errors).
A section of my followme.conf file:
[2000]
context = pstn_inbound
number = 201XXX,5
number = FM1/2000,10
number = FM2/2000,20
number = FM3/2000,30
Here is the relevant section of the macro that calls the FollowMe
app:
exten = s,7,GotoIf(${DIALSTATUS} = NOANSWER?:8:14)
exten = s,8,FollowMe(${STATION_EXTENSION},a)
I've tried different context in my FollowMe configuration file but it
doesn't seem to change anything. Any help would be appreciated.
The app_followme that's in 1.4 right now I don't think ever made use
of
any assets in AstDB, or at least, not what I've coded into it.
BJ
--
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[asterisk-users] Join me on Last.fm!
Hi asterisk-users@lists.digium.com, Add me as a friend on Last.fm so we can share our music taste :) Check out what I'm listening to: http://www.last.fm/user/shina01/?invitedby=shina01tp=ff_tp_b I also sent you a personal note: boo! Signing up is free and takes less than a minute. Just click the link to automatically become my friend. http://www.last.fm/join/?invitedby=shina01tp=ff_tp_b Visit my music profile and leave me a shout! I'll see you around, Sina Owolabi PS: I'm 'shina01' on Last.fm You received this message because someone (Sina Owolabi) who knows you sent you an invitation to join them on Last.fm. Your address was not saved and we will never contact you unsolicited. For more information, see our privacy policy at: http://www.last.fm/help/privacy.php ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Join me on Last.fm!
Classy. On Jan 25, 2008 2:37 PM, Sina Owolabi [EMAIL PROTECTED] wrote: Hi asterisk-users@lists.digium.com, Add me as a friend on Last.fm so we can share our music taste :) Check out what I'm listening to. A personal note from me: boo! Signing up is free and takes less than a minute. Just click here to automatically accept my add. Visit my music profile and leave me a shout! I'll see you around, - Sina Owolabi PS: I'm shina01 on Last.fm. You received this message because someone (Sina Owolabi) who knows you sent you an invitation to join them on Last.fm. Your address was not saved and we will never contact you unsolicited. For more information, see our privacy policy at: http://www.last.fm/help/privacy.php. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erik Anderson http://andersonfam.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel for 1.6-beta1
Is there a minimum zaptel and libpri version for use with 1.6-beta1? Thanks, MC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel for 1.6-beta1
On Friday 25 January 2008 20:03:00 Michael Collins wrote: Is there a minimum zaptel and libpri version for use with 1.6-beta1? zaptel will remain at version 1.4 for the time being, but there is a 1.6-beta1 release of libpri. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Provide a proper link to download Libpri-1.4.3
Hi, I tried to install Libpri-1.4.3 after downloading from sites- www.asterisk.com and www.downloads.digium.com. But in both the case the problem is coming AVC access denied. I am using Fedora core 8. I asked this problem earlier and got advice to disable SELinux. But many people adviced not to do this as it does not require and if it is demanding then there is a bug. I am very much confused. I am very new to Asterisk. Please help me. Thanking you, Preeta pandey Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] External Incomming Call Directed PickUP
My magic orb is on the fritz. Can you give some more info? What extension is ringing? What are you dialing to pick up? What does your conf files look like? I think I might know what the problem is, but I need a little more info. Read core show application Pickup carefully, and then re-read it 3 or 4 more times. It seems odd at first, but then you catch on. You are picking up the calling channel, not the called extension. On Jan 25, 2008 5:28 PM, Fernando Berretta [EMAIL PROTECTED] wrote: Hi, I'm having problems with Directed PickUn and Asterisk 1.4. Directed call pickup **EXT works ok with internal calls which are in the same CONTEXT but,, with calls in which are from other context or incoming calls from IVR this function doesn't work as is pointed in http://bugs.digium.com/view.php?id=11639 I'm using FreePbx 2.3,, and dont know how to solve or workaround this problem Could some one please help me. Best Regards, Fernando ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Avaya 9620 phone using Firmware 2.0.1.34 has working MWI lamp
I just registered an Avaya 9620 set to my Astlinux system (0.47 - Asterisk 1.2.22), using Avaya SIP Firmware version 2.0.1.34. Set [EMAIL PROTECTED] in the sip.conf Found MWI worked immediately. Turned off as expected. Have Fun! Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] External Incomming Call Directed PickUP
Hi, I'm having problems with Directed PickUn and Asterisk 1.4. Directed call pickup **EXT works ok with internal calls which are in the same CONTEXT but,, with calls in which are from other context or incoming calls from IVR this function doesn't work as is pointed in http://bugs.digium.com/view.php?id=11639 I'm using FreePbx 2.3,, and dont know how to solve or workaround this problem Could some one please help me. Best Regards, Fernando ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unprovisioned 7961
I just checked the SIP debug when my 7960 registers and it looks like NAT is enabled and working properly. Does anyone have a 7961 on Asterisk that is going through NAT successfully? -- SIP read from HOME IP ADDRESS:5061: REGISTER sip:TB IP ADDRESS SIP/2.0 Via: SIP/2.0/UDP HOME IP ADDRESS:5061;branch=z9hG4bK740d9e78 From: sip:860001@TB IP ADDRESS;tag=001a6dd2f84c00195c3209da-0ece5aea To: sip:860001@TB IP ADDRESS Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 Date: Fri, 25 Jan 2008 20:20:26 GMT CSeq: 116 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: sip:860001@HOME IP ADDRESS:5061;transport=udp;+sip.instance=urn:uuid:---000 0-001a6dd2f84c;+u.sip!model.ccm.cisco.com=7 Authorization: Digest username=860001,realm=asterisk,uri=sip:TB IP ADDRESS,response=d2b6c69bf9ba5ee5ff808dea90963b64,nonce=5be57786,algor ithm=MD5 Content-Length: 0 Expires: 60 --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to HOME IP ADDRESS : 5061 (NAT) Transmitting (NAT) to HOME IP ADDRESS:5061: SIP/2.0 100 Trying Via: SIP/2.0/UDP HOME IP ADDRESS:5061;branch=z9hG4bK740d9e78;received=HOME IP ADDRESS From: sip:860001@TB IP ADDRESS;tag=001a6dd2f84c00195c3209da-0ece5aea To: sip:860001@TB IP ADDRESS Call-ID: [EMAIL PROTECTED] CSeq: 116 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:860001@TB IP ADDRESS Content-Length: 0 --- Transmitting (NAT) to HOME IP ADDRESS:5061: SIP/2.0 200 OK Via: SIP/2.0/UDP HOME IP ADDRESS:5061;branch=z9hG4bK740d9e78;received=HOME IP ADDRESS From: sip:860001@TB IP ADDRESS;tag=001a6dd2f84c00195c3209da-0ece5aea To: sip:860001@TB IP ADDRESS;tag=as77362809 Call-ID: [EMAIL PROTECTED] CSeq: 116 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 60 Contact: sip:860001@HOME IP ADDRESS:5061;transport=udp;expires=60 Date: Fri, 25 Jan 2008 20:20:26 GMT Content-Length: 0 On 1/25/08 2:54 PM, Gregory Wong [EMAIL PROTECTED] wrote: Thanks Chad. This config seemed to have worked a bit. I don't get the Unprovisioned or Error Verifying Config Info messages anymore. However, the phone sits at Registering and will never register. I took a look at the sip debug and I see the below messages. Do I need to enable NAT in the SEP.cnf.xml file since I am behind NAT? I know my 7960 config file has natEnabled = 1. Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms -- SIP read from MY HOME IP ADDRESS:49157: REGISTER sip:TB IP ADDRESS SIP/2.0 Via: SIP/2.0/UDP MY HOME IP ADDRESS:1140;branch=z9hG4bK48e89c16 From: sip:86003@TB IP ADDRESS;tag=0018195aa6770003efaf5095-54a486b0 To: sip:86003@TB IP ADDRESS Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 Date: Mon, 08 Oct 2007 23:42:08 GMT CSeq: 101 REGISTER User-Agent: Cisco-CP7961G/8.3.0 Contact: sip:86003@HOME IP ADDRESS:1140;transport=udp;+sip.instance=urn:uuid:---000 0-0018195aa677;+u.sip!model.ccm.cisco.com=30018 Supported: (null),X-cisco-xsi-6.0.2 Content-Length: 0 Reason: SIP;cause=200;text=cisco-alarm:25 Name=SEP0018195AA677 Load=SIP41.8-3-3SR2S Last=initialized Expires: 3600 --- (14 headers 0 lines) --- Using latest REGISTER request as basis request Sending to HOME IP ADDRESS : 1140 (non-NAT) Transmitting (no NAT) to HOME IP ADDRESS:1140: SIP/2.0 404 Not found Via: SIP/2.0/UDP HOME IP ADDRESS:1140;branch=z9hG4bK48e89c16;received=HOME IP ADDRESS From: sip:86003@TB IP ADDRESS;tag=0018195aa6770003efaf5095-54a486b0 To: sip:86003@TB IP ADDRESS;tag=as1886ecd1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 On 1/25/08 10:29 AM, Chad Osmond [EMAIL PROTECTED] wrote: Try this configuration file... http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configu ration+files+for+SIP#Cisco7961with833SR2ConfigurationExamples Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Wong Sent: Friday, January 25, 2008 6:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Unprovisioned 7961 Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says Error Verifying Config Info. I have read quite a bit on this topic (getting 7961's to work with Asterisk and TB) and only came across a few postings where other people encountered this issue but no solution was given. I have checked the SEP.cnf.xml file for the phone and everything seems to be right. I even tried to remove some parts of the code as people suggested but no luck. I already have a 7960 on TB so I know that TFTP is working correctly. Any ideas on how I can get this to work would be much appreciated. Thank.
Re: [asterisk-users] Need sample configuration files for sipura/linksys ata
I have emailed Linksys about this, and they have not answered. I have figured out much of how to do this, despite Linksys not being any help. My only remaining issue is how to configure the PSTN line on a SPA3102. Try a file like this (example info included); flat-profile Upgrade_EnableYes/Upgrade_Enable Resync_On_Resetyes/Resync_On_Reset Resync_Random_Delay30/Resync_Random_Delay Resync_Periodic1200/Resync_Periodic Resync_Error_Retry_Delay300/Resync_Error_Retry_Delay Forced_Resync_Delay3600/Forced_Resync_Delay Resync_From_SIPyes/Resync_From_SIP Resync_After_Upgrade_Attemptyes/Resync_After_Upgrade_Attempt Resync_Fails_On_FNFno/Resync_Fails_On_FNF Line_Enable_1_yes/Line_Enable_1_ /flat-profile Notice the Line_Enable_1_. I don't have my PAP2-NA yet, but according to the information I found, line two would be Line_Enable_2_. The lines have a _1_ or _2_ etc etc. Remember to end each line as above. I have mine setup so it downloads the first profile (Profile_Rule) from tftp. That then loads the 3102 with the http site for the rest of the configuration. You can use variables in your URL/TFTP line as well. A $MA will send the MAC address of the adapter, and $PSN is the model number (such as 3102) and $SN is the devices serial number. If you figure out how to specify settings for the PSTN line, please share it with the list. Tim Johnson Quoting Gopal krishnan [EMAIL PROTECTED]: Hi, Try this http://www.kcip.com/support/pap2uk.html On Jan 25, 2008 4:18 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unprovisioned 7961
Thanks Chad. This config seemed to have worked a bit. I don't get the Unprovisioned or Error Verifying Config Info messages anymore. However, the phone sits at Registering and will never register. I took a look at the sip debug and I see the below messages. Do I need to enable NAT in the SEP.cnf.xml file since I am behind NAT? I know my 7960 config file has natEnabled = 1. Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms -- SIP read from MY HOME IP ADDRESS:49157: REGISTER sip:TB IP ADDRESS SIP/2.0 Via: SIP/2.0/UDP MY HOME IP ADDRESS:1140;branch=z9hG4bK48e89c16 From: sip:86003@TB IP ADDRESS;tag=0018195aa6770003efaf5095-54a486b0 To: sip:86003@TB IP ADDRESS Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 Date: Mon, 08 Oct 2007 23:42:08 GMT CSeq: 101 REGISTER User-Agent: Cisco-CP7961G/8.3.0 Contact: sip:86003@HOME IP ADDRESS:1140;transport=udp;+sip.instance=urn:uuid:---000 0-0018195aa677;+u.sip!model.ccm.cisco.com=30018 Supported: (null),X-cisco-xsi-6.0.2 Content-Length: 0 Reason: SIP;cause=200;text=cisco-alarm:25 Name=SEP0018195AA677 Load=SIP41.8-3-3SR2S Last=initialized Expires: 3600 --- (14 headers 0 lines) --- Using latest REGISTER request as basis request Sending to HOME IP ADDRESS : 1140 (non-NAT) Transmitting (no NAT) to HOME IP ADDRESS:1140: SIP/2.0 404 Not found Via: SIP/2.0/UDP HOME IP ADDRESS:1140;branch=z9hG4bK48e89c16;received=HOME IP ADDRESS From: sip:86003@TB IP ADDRESS;tag=0018195aa6770003efaf5095-54a486b0 To: sip:86003@TB IP ADDRESS;tag=as1886ecd1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 On 1/25/08 10:29 AM, Chad Osmond [EMAIL PROTECTED] wrote: Try this configuration file... http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configu ration+files+for+SIP#Cisco7961with833SR2ConfigurationExamples Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Wong Sent: Friday, January 25, 2008 6:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Unprovisioned 7961 Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says Error Verifying Config Info. I have read quite a bit on this topic (getting 7961's to work with Asterisk and TB) and only came across a few postings where other people encountered this issue but no solution was given. I have checked the SEP.cnf.xml file for the phone and everything seems to be right. I even tried to remove some parts of the code as people suggested but no luck. I already have a 7960 on TB so I know that TFTP is working correctly. Any ideas on how I can get this to work would be much appreciated. Thank. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] adding additional volume to console/dsp
Is there a setting that can add additional volume to the Console/Dsp output? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding difficulty in installing Asterisk
On Fri, Jan 25, 2008 at 06:48:21PM +, Deepak Naidu wrote: Before installing ensure selinux is disabled. This should not be needed. If it is, it is a bug. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding difficulty in installing Asterisk
Tzafrir Cohen wrote: On Fri, Jan 25, 2008 at 02:08:02PM +0530, [EMAIL PROTECTED] wrote: Hi, I am using Fedora and I tried to install Libpri-1.4.3 but now I am getting error AVC access denied. Its saying I need to disable SELinux protection. I do not know what to do. Please help me out. Again, what distrbution and vversion of it do you use? Just set the SELINUX=enforcing to SELINUX=disabled in /etc/selinux/config. Reboot and this will disable selinux completely. Then try again. Good luck. Greetings, Joris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding difficulty in installing Asterisk
Before installing ensure selinux is disabled. Check the link below to understand Selinux in Redhat/Fedora. http://www.redhat.com/docs/manuals/enterprise/RHEL-5-manual/Deployment_Guide-en-US/ch-selinux.html Check below link to disable selinux in Fedora, or google around for ur version of fedora. http://docs.fedoraproject.org/selinux-faq-fc3/ -- Deepak [EMAIL PROTECTED] wrote: Hi Dave, I did make clean and then make. But then when I am giving make install its giving error AVC access denied. I am using Fedora. What may be the problem? Help me.. Thanking you, Preeta Pandey -Original Message- From: [EMAIL PROTECTED] on behalf of Dave Cotton Sent: Fri 1/25/2008 1:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Finding difficulty in installing Asterisk On Friday 25 January 2008 05:25:57 Lyle Giese wrote: You need to do a 'make' before the 'make install'. make install will do all that is necessary to install a program including making any files necessary. -- Dave Cotton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Support the World Aids Awareness campaign this month with Yahoo! for Good___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding difficulty in installing Asterisk
On Fri, Jan 25, 2008 at 02:08:02PM +0530, [EMAIL PROTECTED] wrote: Hi, I am using Fedora and I tried to install Libpri-1.4.3 but now I am getting error AVC access denied. Its saying I need to disable SELinux protection. I do not know what to do. Please help me out. Again, what distrbution and vversion of it do you use? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need sample configuration files for sipura/linksys ata
Hi, Try this http://www.kcip.com/support/pap2uk.html On Jan 25, 2008 4:18 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Home use of asterisk
I have found many neat scripts for my home asterisk on the wiki and elsewhere - and we really like it. But there are a couple of things I'd still like to find. And if anyone has some favorites that think think are great for a home with 2 adults and 3 kids (4 phones) 2 cell phones I'd like to hear about them also. What I'd like to find are some scripts that i could modify that use some of the features the phone company provides. Most phone companies have 3 way calling features with only one line. but you have to pass hookflash back to the phone line to get the second dial tone. How would I make use of this feature? Also since I haven't got incoming and sending faxes thru asterisk to be reasonably reliable - and the fax machine that we have can listen on the line for a few seconds - I would like to put it on an incoinf extension - outside of asterisk then when asterisk fax detects it sends 123 or whatever the code the fax machine listens for to start receiving back out the phone line so the fax machine will kick in and do its job. Also if there are any features that would help integrate the cell phones with asterisk - I'd like to hear about those also Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding difficulty in installing Asterisk
It appears as though SELinux is preventing you from moving forward. Perform the following to disable SELinux. cd /etc/selinux vi config change enabled to disabled write your changes reboot Bryan M. Johns Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 Support: [EMAIL PROTECTED] http://www.sheltonjohns.com On Jan 25, 2008, at 3:44 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Dave, I did make clean and then make. But then when I am giving make install its giving error AVC access denied. I am using Fedora. What may be the problem? Help me.. Thanking you, Preeta Pandey -Original Message- From: [EMAIL PROTECTED] on behalf of Dave Cotton Sent: Fri 1/25/2008 1:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Finding difficulty in installing Asterisk On Friday 25 January 2008 05:25:57 Lyle Giese wrote: You need to do a 'make' before the 'make install'. make install will do all that is necessary to install a program including making any files necessary. -- Dave Cotton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Script for seeding polycom phones with an extension directory
Hello List,
Not sure if this will be helpful but I made changes to the original
Cisco directory.php.txt script and applied them for use on the Polycom
phones. This will create an extension directory and alphabetize it
based on the sip registrations you have setup in sip.conf. Note that
this only seeds the phones and does not synchronize them. Anyway
thought it might save people some time. To run do: php scriptname
/home/polycom/-directory.xml.
?
header(Content-type: text/xml);
header(Connection: close);
header(Expires: -1);
// location of asterisk config files
$location = /etc/asterisk/;
// parse sip.conf
$sip_array = parse_ini_file($location.sip.conf, true);
while ($v = current($sip_array))
{ if (isset($v['name']))
{ $directory[] = fn. $v['name']./fn\n.
ct.key($sip_array)./ct\n;
}
next($sip_array);
}
sort($directory);
echo directory\n;
echo item_list\n;
foreach ($directory as $v) {
echo item\n;
echo $v;
echo /item\n;
}
echo /item_list\n;
echo /directory\n;
?
Roy Anciso
Director of Technology
Manistee Intermediate School District
1710 Merkey Road
Manistee, MI 49660
Ph: 231-723-4264
Fx: 231-723-1690
[EMAIL PROTECTED]
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[asterisk-users] Disable IAX2 call path optimization
I have a call coming in from Asterisk-A going to Asterisk-B where it's determined that the called party is in fact yet another number in Asterisk-A so a new call is created from B to A and the two calls bridged (by Asterisk) at Asterisk-B. Originating Caller == Asterisk-A == Asterisk-B == Asterisk-A Now, what happens is that in my case both A and B are on the same network and therefore Asterisk-A apparently optimizes the round-trip to Asterisk-B out and the original caller talks directly to the extension hosted in Asterisk-A without the call path going the round-trip to Asterisk-B. Is it possible to prevent this optimization from happening? Any way to control if it happens at all, or can it be selected on per-call basis somehow? Can I find anywhere more details of call path optimization and it's configuration, use, functionality and behaviour? tnx, Baldvin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with FollowMe
I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed
the WIKI page on setting it up but I can't seem to get it to work.
Here is my Asterisk version:
pbx1*CLI core show version
Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on
2008-01-10
12:08:48 UTC
Here is a log of when the FollowMe is being called:
NOTE: I've tried to use the AstDB as the WIKI described but when I
looked at the code it didn't seem like that would work so I've hard
coded a phone # for now. But I would prefer to use the AstDB if that
is a workable solution.
-- Executing [EMAIL PROTECTED]:8] FollowMe(Zap/
1-1,2000|a) in new stack
-- Zap/1-1 Playing 'vm-rec-name' (language 'en')
-- Zap/1-1 Playing 'beep' (language 'en')
-- x=0, open writing: /var/spool/asterisk/followme.1201278830.10
format: sln, 0x81c0218
-- User ended message by pressing #
-- Zap/1-1 Playing 'auth-thankyou' (language 'en')
-- Zap/1-1 Playing 'followme/pls-hold-while-try' (language 'en')
-- Started music on hold, class 'default', on channel 'Zap/1-1'
[Jan 25 11:34:32] NOTICE[23725]: chan_local.c:571 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel
[Jan 25 11:34:32] WARNING[23725]: app_followme.c:863 findmeexec:
Unable to allocate a channel for Local/FM1/[EMAIL PROTECTED] cause:
Unknown
[Jan 25 11:34:32] NOTICE[23725]: chan_local.c:571 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel
[Jan 25 11:34:32] WARNING[23725]: app_followme.c:863 findmeexec:
Unable to allocate a channel for Local/FM2/[EMAIL PROTECTED] cause:
Unknown
[Jan 25 11:34:32] NOTICE[23725]: chan_local.c:571 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel
[Jan 25 11:34:32] WARNING[23725]: app_followme.c:863 findmeexec:
Unable to allocate a channel for Local/FM3/[EMAIL PROTECTED] cause:
Unknown
(NOTE: This log was taken just prior to making the changes indicated
below in my Followme.conf file. But the changes reported the same logs
lines, simply with different values (i.e. same errors).
A section of my followme.conf file:
[2000]
context = pstn_inbound
number = 201XXX,5
number = FM1/2000,10
number = FM2/2000,20
number = FM3/2000,30
Here is the relevant section of the macro that calls the FollowMe app:
exten = s,7,GotoIf(${DIALSTATUS} = NOANSWER?:8:14)
exten = s,8,FollowMe(${STATION_EXTENSION},a)
I've tried different context in my FollowMe configuration file but it
doesn't seem to change anything. Any help would be appreciated.
Thanks,
Mike
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Re: [asterisk-users] Unprovisioned 7961
Well it would seem that Cisco chose not to make their SEP_MAC.cnf.xml format standard across all of their phone models. I have had similar issues getting various models to work. It has made it a challenge, for no obvious good reason IMO. I have pasted a SEP_MAC.cnf.xml that I use for a 7941G-GE. Give it a try if you like, or just use it for comparison Glenn ?xml version=1.0 encoding=UTF-8 standalone=yes? device xsi:type=axl:XIPPhone ctiid=1566023366 deviceProtocolSIP/deviceProtocol sshUserIdroot/sshUserId sshPasswordroot/sshPassword devicePool dateTimeSetting dateTemplateM/D/Ya/dateTemplate timeZoneEastern Standard/Daylight Time/timeZone ntps ntp name10.10.30.10/name ntpModeUnicast/ntpMode /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeName10.10.30.10/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies backupProxy/backupProxy backupProxyPort/backupProxyPort emergencyProxy/emergencyProxy emergencyProxyPort/emergencyProxyPort outboundProxy/outboundProxy outboundProxyPort/outboundProxyPort registerWithProxytrue/registerWithProxy /sipProxies sipCallFeatures cnfJoinEnabledtrue/cnfJoinEnabled callForwardURIx--serviceuri-cfwdall/callForwardURI callPickupURIx-cisco-serviceuri-pickup/callPickupURI callPickupListURIx-cisco-serviceuri-opickup/callPickupListURI callPickupGroupURIx-cisco-serviceuri-gpickup/callPickupGroupURI meetMeServiceURIx-cisco-serviceuri-meetme/meetMeServiceURI abbreviatedDialURIx-cisco-serviceuri-abbrdial/abbreviatedDialURI rfc2543Holdfalse/rfc2543Hold callHoldRingback2/callHoldRingback localCfwdEnabletrue/localCfwdEnable semiAttendedTransfertrue/semiAttendedTransfer anonymousCallBlock2/anonymousCallBlock callerIdBlocking2/callerIdBlocking dndControl1/dndControl remoteCcEnabletrue/remoteCcEnable /sipCallFeatures sipStack sipInviteRetx6/sipInviteRetx sipRetx10/sipRetx timerInviteExpires180/timerInviteExpires timerRegisterExpires3600/timerRegisterExpires timerRegisterDelta5/timerRegisterDelta timerKeepAliveExpires120/timerKeepAliveExpires timerSubscribeExpires120/timerSubscribeExpires timerSubscribeDelta5/timerSubscribeDelta timerT1500/timerT1 timerT24000/timerT2 maxRedirects70/maxRedirects remotePartyIDfalse/remotePartyID userInfoNone/userInfo /sipStack autoAnswerTimer1/autoAnswerTimer autoAnswerAltBehaviorfalse/autoAnswerAltBehavior autoAnswerOverridetrue/autoAnswerOverride transferOnhookEnabledfalse/transferOnhookEnabled enableVadfalse/enableVad preferredCodecnone/preferredCodec dtmfAvtPayload101/dtmfAvtPayload dtmfDbLevel3/dtmfDbLevel dtmfOutofBandavt/dtmfOutofBand alwaysUsePrimeLinefalse/alwaysUsePrimeLine alwaysUsePrimeLineVoiceMailtrue/alwaysUsePrimeLineVoiceMail kpml3/kpml natEnabled/natEnabled natAddress/natAddress phoneLabelExt 3105/phoneLabel stutterMsgWaiting0/stutterMsgWaiting callStatsfalse/callStats silentPeriodBetweenCallWaitingBursts10/silentPeriodBetweenCallWaitingBurs ts disableLocalSpeedDialConfigfalse/disableLocalSpeedDialConfig startMediaPort16384/startMediaPort stopMediaPort32766/stopMediaPort sipLines line button=1 featureID9/featureID featureLabel3105/featureLabel proxy10.10.30.10/proxy port5060/port name3105/name displayName3105/displayName autoAnswer autoAnswerEnabled1/autoAnswerEnabled /autoAnswer callWaiting3/callWaiting authName3105/authName authPassword3105/authPassword sharedLinefalse/sharedLine messageWaitingLampPolicy3/messageWaitingLampPolicy messagesNumber*97/messagesNumber ringSettingIdle4/ringSettingIdle ringSettingActive5/ringSettingActive contact3105/contact forwardCallInfoDisplay callerNametrue/callerName callerNumberfalse/callerNumber redirectedNumberfalse/redirectedNumber dialedNumbertrue/dialedNumber /forwardCallInfoDisplay /line line
Re: [asterisk-users] Intercepting DTMF to initiate Voice Drop
From: Raj Jain - Friday, January 25, 2008 10:07 AM
I'm trying to implement a Voice Drop service within Asterisk
dial-plan. The service is supposed to work as following:
1. A initiates a call to B
2. The call is answered by B's answering machine
3. A hears the answering machine's greeting and the recording beep
4. A speaks a few words into the recording to personalize the message
5. A presses some DTMF keys (say, '##') to initiate Voice Drop
6. PBX intercepts DTMF and starts playing a prerecorded
announcement to B
7. A is released from the call as soon as the Voice Drop is initiated
8. PBX releases the call to B at the end of the announcement
Any thoughts, ideas?
After talking with B, A could transfer the call to an extension such as
123 with a dial plan something like:
Exten = 123,1,Playback(file)
Exten = 123,n,Playback(file)
Exten = 123,n,hangup
A will need to be able to transfer outgoing calls ('T' option).
Don Pobanz
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Re: [asterisk-users] Unprovisioned 7961
Try this configuration file... http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configu ration+files+for+SIP#Cisco7961with833SR2ConfigurationExamples Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Wong Sent: Friday, January 25, 2008 6:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Unprovisioned 7961 Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says Error Verifying Config Info. I have read quite a bit on this topic (getting 7961's to work with Asterisk and TB) and only came across a few postings where other people encountered this issue but no solution was given. I have checked the SEP.cnf.xml file for the phone and everything seems to be right. I even tried to remove some parts of the code as people suggested but no luck. I already have a 7960 on TB so I know that TFTP is working correctly. Any ideas on how I can get this to work would be much appreciated. Thank. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum Paging Group Size?
I have done a page with at least a hundred phones before. It took about a full second for the mysql script to run and all the phones to join the conference, but worked fine. We typically only page 60 phones at once. In the coming months, I will be attempting a page with 250 phones. --- Forrest Beck http://www.shift8.biz On Jan 25, 2008, at 3:47 AM, George Pajari wrote: Has anyone experience with (or an educated guess of) the largest paging group that can be supported by the Page() command? We have an installation coming up with 110 phones -- any hope to page this entire facility? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum Paging Group Size?
I can confirm that. 110 phones should not be a problem. We've done paging groups at that size. The only noticeable issue is the delay. When the initiator starts to speak, there may be up to a 1 second delay for all the phones to receive the audio. However, you probably wouldn't notice that unless you had all the phones in a single room for testing... :-) In production with one phone in each room in scattered locations, it should not be an issue. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 - Original Message - From: Forrest Beck [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 25, 2008 9:32:59 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] Maximum Paging Group Size? I have done a page with at least a hundred phones before. It took about a full second for the mysql script to run and all the phones to join the conference, but worked fine. We typically only page 60 phones at once. In the coming months, I will be attempting a page with 250 phones. --- Forrest Beck http://www.shift8.biz On Jan 25, 2008, at 3:47 AM, George Pajari wrote: Has anyone experience with (or an educated guess of) the largest paging group that can be supported by the Page() command? We have an installation coming up with 110 phones -- any hope to page this entire facility? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Intercepting DTMF to initiate Voice Drop
Hi, I'm trying to implement a Voice Drop service within Asterisk dial-plan. The service is supposed to work as following: 1. A initiates a call to B 2. The call is answered by B's answering machine 3. A hears the answering machine's greeting and the recording beep 4. A speaks a few words into the recording to personalize the message 5. A presses some DTMF keys (say, '##') to initiate Voice Drop 6. PBX intercepts DTMF and starts playing a prerecorded announcement to B 7. A is released from the call as soon as the Voice Drop is initiated 8. PBX releases the call to B at the end of the announcement To acheive this I need to intercept DTMF in the middle of a call and initiate an action based on that. I couldn't find an option in the Dial() application to break out of it on receipt of a particular DTMF sequence. Does the Dial() application support such a capability? I've tried the 'G' option in the Dial() application but that splits the call as soon as it is answered, whereas, I need to split the call after it is established based on a DTMF stimulus. Are there any other ways of accomplishing this goal? Any thoughts, ideas? Thank you, Raj Jain mailto:rj2807 at gmail dot com sip:rjain at iptel dot org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with FollowMe
Mike Coakley wrote:
I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed
the WIKI page on setting it up but I can't seem to get it to work.
Here is my Asterisk version:
pbx1*CLI core show version
Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on
2008-01-10
12:08:48 UTC
Here is a log of when the FollowMe is being called:
NOTE: I've tried to use the AstDB as the WIKI described but when I
looked at the code it didn't seem like that would work so I've hard
coded a phone # for now. But I would prefer to use the AstDB if that
is a workable solution.
-- Executing [EMAIL PROTECTED]:8] FollowMe(Zap/
1-1,2000|a) in new stack
-- Zap/1-1 Playing 'vm-rec-name' (language 'en')
-- Zap/1-1 Playing 'beep' (language 'en')
-- x=0, open writing: /var/spool/asterisk/followme.1201278830.10
format: sln, 0x81c0218
-- User ended message by pressing #
-- Zap/1-1 Playing 'auth-thankyou' (language 'en')
-- Zap/1-1 Playing 'followme/pls-hold-while-try' (language 'en')
-- Started music on hold, class 'default', on channel 'Zap/1-1'
[Jan 25 11:34:32] NOTICE[23725]: chan_local.c:571 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel
[Jan 25 11:34:32] WARNING[23725]: app_followme.c:863 findmeexec:
Unable to allocate a channel for Local/FM1/[EMAIL PROTECTED] cause:
Unknown
[Jan 25 11:34:32] NOTICE[23725]: chan_local.c:571 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel
[Jan 25 11:34:32] WARNING[23725]: app_followme.c:863 findmeexec:
Unable to allocate a channel for Local/FM2/[EMAIL PROTECTED] cause:
Unknown
[Jan 25 11:34:32] NOTICE[23725]: chan_local.c:571 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel
[Jan 25 11:34:32] WARNING[23725]: app_followme.c:863 findmeexec:
Unable to allocate a channel for Local/FM3/[EMAIL PROTECTED] cause:
Unknown
(NOTE: This log was taken just prior to making the changes indicated
below in my Followme.conf file. But the changes reported the same logs
lines, simply with different values (i.e. same errors).
A section of my followme.conf file:
[2000]
context = pstn_inbound
number = 201XXX,5
number = FM1/2000,10
number = FM2/2000,20
number = FM3/2000,30
Here is the relevant section of the macro that calls the FollowMe app:
exten = s,7,GotoIf(${DIALSTATUS} = NOANSWER?:8:14)
exten = s,8,FollowMe(${STATION_EXTENSION},a)
I've tried different context in my FollowMe configuration file but it
doesn't seem to change anything. Any help would be appreciated.
The app_followme that's in 1.4 right now I don't think ever made use of
any assets in AstDB, or at least, not what I've coded into it.
BJ
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Re: [asterisk-users] Disable IAX2 call path optimization
You are usinfg sip or iax ? Its possible to prevent in both cases for sip under peer definition you can put canreinvite=no and in iax2 you can put transfer=no for asterisk 1.4 or notranser=yes in asterisk 1.2 .. Search for this on voip-info.org wiki for more info . On Jan 25, 2008 7:03 PM, [EMAIL PROTECTED] wrote: I have a call coming in from Asterisk-A going to Asterisk-B where it's determined that the called party is in fact yet another number in Asterisk-A so a new call is created from B to A and the two calls bridged (by Asterisk) at Asterisk-B. Originating Caller == Asterisk-A == Asterisk-B == Asterisk-A Now, what happens is that in my case both A and B are on the same network and therefore Asterisk-A apparently optimizes the round-trip to Asterisk-B out and the original caller talks directly to the extension hosted in Asterisk-A without the call path going the round-trip to Asterisk-B. Is it possible to prevent this optimization from happening? Any way to control if it happens at all, or can it be selected on per-call basis somehow? Can I find anywhere more details of call path optimization and it's configuration, use, functionality and behaviour? tnx, Baldvin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
