Re: [Asterisk-Users] Provisioning lines 5 and 6 via TFTP
On Apr 22, 2005, at 2:13 AM, C F wrote: Can you please post your .cnf files? On 4/21/05, Robert Goodyear [EMAIL PROTECTED] wrote: Has anyone experienced a problem provisioning lines 5 and 6 of a Cisco 7960 via a SIPx.CNF over TFTP? I'm going to try Ron Wellsted's suggestion re the .CNF filesize limitation bug and then I'll report back to the list my findings. If that doesn't work I'll share my files. /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Recommended Linux Dist. for Asterisk
Using RH 9 with * Regards, Umair Bari David Choo wrote: We used gentoo internally. I also have * running on CentOS, RHEL. Best Regards, == David Choo Systems Engineer Business Technology Division "Engineered for Changing Businesses" Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-68487806 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. Michael George [EMAIL PROTECTED] a.com To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 21/04/2005 10:31 Re: [Asterisk-Users] Recommended PMLinux Dist. for Asterisk Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com On Wed, Apr 20, 2005 at 10:26:33PM -0500, Paul Shiflet wrote: I'm trying to find out what flavor of Linux people are choosing for their asterisk boxes. I have been using RH, but i'd like to try some different ones. It seems that RH is the common denominator in this rash of line noise problems. So some suggestions for what dist to use would be great. We use gentoo. Many people would not go that route, but we use that on our servers because when we are ready to update it, we can do so with less pain than with RHL/Fedora and SuSE, etc. The updates of the latter usually go okay, but there comes the time when we need to change major releases and that should be done with a clean reinstall. Now, with * you don't really need to do any changing as it will just sit there and work for the most part. However, since we have gentoo in many of our systems, we just stick with that. The ports in gentoo stay pretty current and it's worked fine for us. YMMV, and as I said above, gentoo is probably not the route for many who have little linux experience. -- -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 mailing list Asterisk-Users@lists.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] QOS Routers
On Apr 22, 2005, at 2:54 PM, Jay Milk wrote: Sveasoft is useless -- use hyperWRT instead. -Original Message- From: Gregory Wiktor - ADCom Corp. [mailto:[EMAIL PROTECTED] Sent: Friday, April 22, 2005 4:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] QOS Routers How about a linksys wrt54g with sveasoft firmware? Has some shaping and many other nice features... Jay: can you elaborate on your standpoint on the svea firmware? thx /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Hotel billing in IPSwitchBoard
I am currently working on implementing Hotel Billing in IPSwitchBoard. The idea is that a receptionist in a hotel can just right click an extension button and choose Account; IPS will now calculate the call charges made from that extension and show all calls and charges on a form. The receptionist now has the option to close the account which will reset the account. I will add a table for editing call charges, and there will be a possibility to add a fee for connection charges and also an option to charge calls per xx seconds and to add/subtract a percentage to all calls. I will add a family/key to the asterisk database to indicate if the extension is closed, this way you can stop outgoing calls from being made from a closed extension by checking the dial plan. Please let me know if there are any other features you would like to see in IPSwitchBoard. Thorben Download IPSwitchBoard for free here: http://ipswitchboard.thorben.dk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fwd: [Asterisk-Users] Asterisk transcoding
--- Weitergeleitete Nachricht / Forwarded Message --- Date: Fri, 22 Apr 2005 14:12:04 +0200 (MEST) From: Georg Natsikos [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk transcoding I would like to learn more over the transcoding function with asterisk. How exactly works asterisk, in order to transcoding. Where I can get exactly informations? If asterisk transcodes, for example ilbc to gsm, as I can see which (ilbc) rtp-packet becomes which (gsm) rtp-packet? would be very grateful for assistance -- +++ GMX - Die erste Adresse für Mail, Message, More +++ 1 GB Mailbox bereits in GMX FreeMail http://www.gmx.net/de/go/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- +++ NEU: GMX DSL_Flatrate! Schon ab 14,99 EUR/Monat! +++ GMX Garantie: Surfen ohne Tempo-Limit! http://www.gmx.net/de/go/dsl ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IAX help
On 23/04/05, Michael DiMartino [EMAIL PROTECTED] wrote:
3. Extensions.conf (telx-NY17S)
;Extentions at telx-nyc
exten = _7XXX,1,Dial(IAX2/telx-nyc/${EXTEN})
exten = _7XXX,1,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN})
where username:password is the credientials you need to authenticate
with the other server.
The username/secret in iax2.conf is for inbound, not for outbound calls.
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Quadbri bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ?
I have eight MSN at home. Six are handled by Asterisk. The two remaining are handled by an external ISDN modem which is connected to my HylaFax on another machine. Asterisk and ISDN modem are plugged into the same NT1. That works fine for me... Works for me too. We have an old fax machine sitting on the same NT1 as asterisk. In asterisk I ignored the MNS by setting the line exten = my_fax_msn,1,wait(30) -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Hotel billing in IPSwitchBoard
This seems to be exactly the application I was looking for :-). Since I'm working on a project where accounting and billing (http and voip traffic) is an issue, I'm glad to read that there will be a solution within a reasonable GUI. While dealing with squid and the great Squid2MYSQL script - used for Accounting and Billing - from Eugene V. Chernyshev [EMAIL PROTECTED], I wasn't shure about the udp traffic. A time based Billing seems to be a good solution. Keep on going with your great work on IPSwitchboard. Guido Hecken I am currently working on implementing Hotel Billing in IPSwitchBoard. The idea is that a receptionist in a hotel can just right click an extension button and choose Account; IPS will now calculate the call charges made from that extension and show all calls and charges on a form. The receptionist now has the option to close the account which will reset the account. I will add a table for editing call charges, and there will be a possibility to add a fee for connection charges and also an option to charge calls per xx seconds and to add/subtract a percentage to all calls. I will add a family/key to the asterisk database to indicate if the extension is closed, this way you can stop outgoing calls from being made from a closed extension by checking the dial plan. Please let me know if there are any other features you would like to see in IPSwitchBoard. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Failed to authenticate
HI,all!
I have setted up an Asterisk server on my linux.My sip.conf and extensions.conf configurations like these:
sip.conf
[general]port=5060disallow=allallow=alawallow=ulawallow=gsmnat=yes
dtmfmode=rfc2833canreinvite=no
context=defaulttos=0x18dbname=asteriskdbhost=localhostdbuser=asteriskdbpass=password
extensions.conf
[general]static=yeswriteprotect=no
[globals]CONSOLE=Console/dsp
[local]
exten = _X.,1,Dial(SIP/${EXTEN},20,t)exten = _X.,2,Hangup
[default]include = demoinclude = local
I have also setted callidnum 1000-1010 in mysql database.First,it can dial out and receive a call well.(in internal) then I alter callidnum 1000 to
1000.It can registered successfully and it can receive a call ,but it cannot dial out .There are some words in my asterisk console:"Failed to authenticate user "aaa" sip:[EMAIL PROTECTED]; tag=164262242".So,I tried change callidnum to 1000, it works. I don't know what happen.Can anybody tell me what's the matter ? thanks!
Do You Yahoo!?
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[Asterisk-Users] ast_expr.y:243 to_integer:Overflow
hi,all! Can anybody tell me what's the matter? thanks!!! Do You Yahoo!? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] chan_sip.c:7174 handle_request : Failed to authenticate user
Hi,all ! My asterisk's console appear some words : "chan_sip.c:7174 handle_request : Failed to authenticate user "top" sip:1002:@10.0.0.1 tag=169447308" . Can anybody tell me what cause it ? thanks!!! Do You Yahoo!? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] usb phone(AU-100) and usb phone adapter(TJ560B)
My notebook has three USB ports. I would like to use usb-phone(AU-100) and usb-analogphone-adapter(TJ560B) using 'wcusb','wcfxs' and 'zap/1' and 'zap/2' on CVS-v1-0-03/05/05 on FC3(2.6.11-1.14_FC3). I could not make /dev/zap/1, /dev/zap/2 for usb devices. How should I do? Do I need X100P type(PCI-bus) interface for zap channel for notebook? # lsusb Bus 004 Device 001: ID : Bus 003 Device 002: ID 0d8c:000e C-Media Electronics, Inc. Bus 003 Device 001: ID : Bus 002 Device 003: ID 06e6:c31c Tiger Jet Network, Inc. Bus 002 Device 002: ID 05e3:1205 Genesys Logic, Inc. Afilias Optical Mouse H3003 Bus 002 Device 001: ID : Bus 001 Device 001: ID : # lsmod snd_usb_audio 65153 0 snd_usb_lib13121 1 snd_usb_audio snd_rawmidi28641 1 snd_usb_lib snd_seq_device 8781 1 snd_rawmidi ztdummy 3924 0 wcusb 19616 0 wcfxs 31904 0 zaptel204804 8 ztdummy,wcte11xp,wcusb,wcfxs,wcfxo,wct1xxp,wct4x uhci_hcd 33497 0 ...(snip) # ztcfg ZT_CHANCONFIG failed on channel 2: No such device or address (6) # asterisk -vc .(snip)... [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Automatically generated pseudo channel == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI) == Registered application 'CallingPres' == Manager registered action ZapTransfer == Manager registered action ZapHangup == Manager registered action ZapDialOffhook == Manager registered action ZapDNDon == Manager registered action ZapDNDoff == Manager registered action ZapShowChannels .(snip)... # ls -al /dev .(snip).. drwxr-xr-x 2 rootroot 120 4 23 19:02 zap crw--- 1 rootroot196, 253 4 23 19:02 zaptel crw-rw-rw- 1 rootroot 1, 5 4 24 2005 zero # ls -al /dev/zap drwxr-xr-x 2 root root 120 4 23 19:02 . drwxr-xr-x 11 root root 6240 4 23 19:27 .. crw-rw 1 root root 196, 254 4 23 19:02 channel crw-rw 1 root root 196, 0 4 23 19:02 ctl crw-rw 1 root root 196, 255 4 23 19:02 pseudo crw-rw 1 root root 196, 253 4 23 19:02 timer Regards, Zen ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Dial While on IVR
Title: Dial While on IVR While the call is going into the IVR how can I Dial an extension and get immediately connected interrupting the IVR? Robson ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Hotel billing in IPSwitchBoard
Exactly what I am looking for also. Because we have multiple phones in one villa, I would need the ability to group extensions and produce an overall bill, and I would, of course, need the ability to set the charge rate versus the cost, i.e., the cost is $.02/min, but we might charge $.50/min regardless of destination, a flat fee for all long distance and international. This is so cool. Chris Mason www.anguillaguide.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thorben Jensen Sent: Saturday, April 23, 2005 3:12 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Hotel billing in IPSwitchBoard I am currently working on implementing Hotel Billing in IPSwitchBoard. The idea is that a receptionist in a hotel can just right click an extension button and choose Account; IPS will now calculate the call charges made from that extension and show all calls and charges on a form. The receptionist now has the option to close the account which will reset the account. I will add a table for editing call charges, and there will be a possibility to add a fee for connection charges and also an option to charge calls per xx seconds and to add/subtract a percentage to all calls. I will add a family/key to the asterisk database to indicate if the extension is closed, this way you can stop outgoing calls from being made from a closed extension by checking the dial plan. Please let me know if there are any other features you would like to see in IPSwitchBoard. Thorben Download IPSwitchBoard for free here: http://ipswitchboard.thorben.dk ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Re: QOS Routers
These things are dirt cheap. Are they any good? MARK. Iassen Hristov wrote: Maybe this fits the bill. http://www.gigafast.com/products/product_detail/EE2400-SS.htm It retails for less than $100 Message: 9 Date: Fri, 22 Apr 2005 10:42:20 -0700 From: Max Clark [EMAIL PROTECTED] Subject: [Asterisk-Users] QOS Routers To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. TIA, 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 mailing list Asterisk-Users@lists.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't make my PRI dial out
did this. No joy Ken Godee wrote: I added the line exten = 3701,1,Dial(Zap/g1/19173657597) Unknown Number Plan (0) '19173657597' ] -- Called g1/19173657597 I know we are moving forward. I didn;t get this last time I tried to dial. Try striping the 1 off and dial Dial(Zap/g1/9173657597) ___ Asterisk-Users mailing list Asterisk-Users@lists.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't make my PRI dial out
My circuit is from MCI. They tell me to its and ATT switchtype Andres wrote: I know we are moving forward. I didn;t get this last time I tried to dial. Mark Why don't you try changing your switchtype to national from 4ess in your zapata.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Re: Hotel billing in IPSwitchBoard
Exactly what I am looking for also. Because we have multiple phones in one villa, I would need the ability to group extensions and produce an overall bill, and I would, of course, need the ability to set the charge rate versus the cost, i.e., the cost is $.02/min, but we might charge $.50/min regardless of destination, a flat fee for all long distance and international. This is so cool. Hi Chris Grouping is a good idea, will not be in the first release, but later. There will only be a charge rate in the first release. You can charge depending on the destination. Thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No sound with voicemail and musiconhold?!?
I've no timer configured, that's it. Thank you for your help. -- Antoine Ron Wellsted a écrit : -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Antoine Courouble wrote: Hi! I'am a new user and have problem with sound on a debian sarge. I can't play any sound with musiconhold or voicemail. Sounds on var/lib have good rights and mpg123 is installed. On console asterisk stops in the first playing. Someone have same problem or can help me? Have you compiled and installed zaptel and loaded ztdummy? HTH - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD:519961 Gossiptel:9309811 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBQmlBlktP/KMNOfRbAQIifAgAsKLIJF2os9qw2atyJUqcQN1mKpMyiaRx OX5hhusdEnNYsMQzIUXh0nPyC5uT2R0SUUfWvz61wIxwIi3II1ozfBA9475ifncN iWNDQsomiybcpZxSrwKn3/ZZb4StMIiNKJo36XuAMelyqBq1tnYlPoDV9gyJLzQr 6mqg8+2TELazL5WLtlgMKQN2Grn//haTdRa124v8kWNbFmY1p6hNknjR6PZxIHj8 zRj6+joD8JCidWmR3BrNGB1IqdOkzaNATDR12DU3oub69lbvTd1JMQcvC0/aO7ic xlW85yNfu37APIqbMNdUaHHe/v+q4QJwL02WSTj9NMeMx/Jo9lnRmQ== =sTVF -END PGP 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 mailing list Asterisk-Users@lists.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't make my PRI dial out
Having said that I did it anyway and whadyaknow, it works!! Thanks very much fellas for your help. Mark Mark Phillips wrote: My circuit is from MCI. They tell me to its and ATT switchtype Andres wrote: I know we are moving forward. I didn;t get this last time I tried to dial. Mark Why don't you try changing your switchtype to national from 4ess in your zapata.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard
Also needed is a way to title and logo the print out so it looks like an invoice. A tempplate would work, and if can use HTML templates that would be easy to customise. Consider making the data a table that is substituted into the html template. Chris Mason www.anguillaguide.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of tgj Sent: Saturday, April 23, 2005 7:55 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard Exactly what I am looking for also. Because we have multiple phones in one villa, I would need the ability to group extensions and produce an overall bill, and I would, of course, need the ability to set the charge rate versus the cost, i.e., the cost is $.02/min, but we might charge $.50/min regardless of destination, a flat fee for all long distance and international. This is so cool. Hi Chris Grouping is a good idea, will not be in the first release, but later. There will only be a charge rate in the first release. You can charge depending on the destination. Thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Re: can't make my PRI dial out
On April 22, 2005 05:35 pm, Mark Phillips wrote: Ext: 1 Cause: Invalid information element contents (100), class = Protocol Error (6) ] -- Processing IE 8 (cs0, Cause) Your zapata.conf does not match your telco's provisioning of the PRI. Contact your switch tech and verify your settings. I happened to notice that your cell number is in the 917 area. Is your PRI provider Verizon? If so, I can send you my config that works for a Verizon PRI. Offhand, I am going to guess that you should *NOT* set caller ID **NAME** -- send only the number. Failing that, don't even set the CID number. Yes, Verizon lets me specify CID number, but not name. - Noah ___ Asterisk-Users mailing list Asterisk-Users@lists.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 7960 won't register as SIP device
It can use DNS if the DNS servers are valid. Can you post your SIP.conf? Didi you configure the phone manually or did you use the cnf files? If you used cnf files can you post those also? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phones and firewalls ...
When you do a sip show peers from the what IP address does it show for the 841? - Original Message - From: Brian Watters [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 12:52 AM Subject: [Asterisk-Users] IP Phones and firewalls ... Hello all, Here is our problem .. IP SIP phones remote .. They will connect to our IP PBX (Asterisk Server) without issue however no voice makes it when anyone answers a phone call made by one of these IP phones. So this means SIP is working but RTP is not, Here is what I currently have on the firewall (http://m0n0.ch/wall). Firewall Rules TCP/UDP * * 192.168.2.253 5060 NAT SIP Protocol UDP * * 192.168.2.253 4569 NAT IAX Protocol UDP * * 192.168.2.253 5036 NAT IAX Protocol UDP * * 192.168.2.253 1 - 2 NAT RTP UDP NAT Rules WAN TCP/UDP 5060 - 5099 192.168.2.253 5060 SIP Protocol WAN UDP 4569 192.168.2.253 4569 IAX2 Protocol WAN UDP 5036 192.168.2.253 5036 IAX Protocol WAN UDP 1 - 2 192.168.2.253 1 - 2 RTP UDP Range IP phones are Sipra 841's and work great when on the same subnet as the * server, this only becomes an issue when offnet and of course outside of the firewall. So I am stumped as to why this does not work .. I have logging turned on for all of the above and see no packets getting dropped .. Anyone there able to shead some light on this .. Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.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] PA168 ip phone setup iax2 to LiveVoip
Can anyone PLEASE help to get a pa168 ip phone connected to livevoip? If I set use service it does not work. If I unset it, it works for a while, then just busy tone. If I set and unset use service, it will again work for a while. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.2 - Release Date: 21/04/2005 ___ 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
Re: [Asterisk-Users] Re: IPswitch: How to use speed dialing?
Has anybody success with speed dialing?
If so, I am sure you can help me to get into this club.
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] if outgoing call fails with provider 1 then auto try provider 2
When someone teminates a call with my softphone to my asterisk server i want asterisk to try provider 1 first and if the call does not go through because the provider is having problems then it will try provider 2 gracefully. I realize that the provider is having problems statement is hard to descrribe. There could be so many problems! For example: 1) an account password got changed. 2) account was accidentally disabled. 3) network problems of any kind. 4) constant busy(I have heard of this happening sometimes?) 5) iax2 show peers show them having a problem. 6) use your imagination, there are lots more I have searched the wiki and other sources and cannot find an extensions.conf example that will accomplish this. If someone can post an exapmle I will add it to the wiki. I am trying to make things as bulletproof as possible. Thanks, Tom There are no real examples that would address your points. The primary reason is that your * can dispatch a call to a provider and the provider will accept that handshaking call. But, if they are having internal call-completion issues, there is no way for you to know that. You could get some sort of busy, dead air, etc. You could probably design some sort of timer-based timeout, but what indication would you use to indicate the call was successful vs unsuccessful? There are several ways to address whether your * is successful in reaching your provider's equipment, but that's about it. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] OctoBRI and 2.6kernel
Hi Guys I am trying to get the Junghanns card to load on Suse 9.3 and tried to get it running on Fedora Core 3 (latest kernels). I have heard from a source here in South Africa that this is about as hard as pulling teeth. Could someone please confirm this for me and if they do have it working properly is it possible to get a guide. I can get the zaptel and qozap to load the card and all the ports and inside asterisk I see the zap channels. But I cannot get a line out or make any incoming calls. Are there some 2.6 tweaks that I need to do in the kernel. Kind Regards Terry Wade Mobile: +27 82 802-5750 Office: +27 11 784-7642 Fax: +27 11 388-0855 Linux is like a Wigwam - No gates, no windows, Apache inside Disclaimer and Confidentiality Warning This message is intended for the addressee only. If you are not the intended recipient of this message, you are notified that any distribution, use of or copying of this communication is strictly prohibited. If you have received the communication in error, please notify the sender immediately. The views and opinions expressed in this message are those of the individual sender of this message and do not necessarily represent the views and opinions of ActiCom. Consequently, ActiCom does not accept responsibility for such views and opinions and this message should not be read as representing the views and opinions of ActiCom without subsequent written confirmation. Each page attached hereto must also be read in conjunction with this disclaimer. ___ Asterisk-Users mailing list Asterisk-Users@lists.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 7960 won't register as SIP device
The DNS servers are valid. I configured the phone via .cnf files. The following are the sip.conf and sipMAC.cnf files. [tycisco] type=friend username=username secret=secret qualify=200 ; Qualify peer is no more than 200ms away nat=yes ;insecure=no host=dynamic; This device registers with us ;defaultip=24.18.147.95 canreinvite=no context=fullaccess dtmfmode=inband ;mailbox=101 disallow=all allow=ulaw allow=alaw allow=g729 .cnf: # SIP Configuration File (start) # Proxy Server proxy1_address: asterisk.mastermindpro.com proxy2_address: proxy3_address: proxy4_address: proxy5_address: proxy6_address: # Line 1 Settings line1_name: tycisco ; Line 1 Extension\User ID line1_displayname: 101 ; Line 1 Display Name line1_authname: username ; Line 1 Registration Authentication line1_password: secret ; Line 1 Registration Password # Line 2 Settings line2_name: ; Line 2 Extension\User ID line2_displayname:; Line 2 Display Name line2_authname: UNPROVISIONED ; Line 2 Registration Authentication line2_password: UNPROVISIONED ; Line 2 Registration Password # Line 3 Settings line3_name: ; Line 3 Extension\User ID line3_displayname:; Line 3 Display Name line3_authname: UNPROVISIONED ; Line 3 Registration Authentication line3_password: UNPROVISIONED ; Line 3 Registration Password # Line 4 Settings line4_name: ; Line 4 Extension\User ID line4_displayname:; Line 4 Display Name line4_authname: UNPROVISIONED ; Line 4 Registration Authentication line4_password: UNPROVISIONED ; Line 4 Registration Password # Line 5 Settings line5_name: ; Line 5 Extension\User ID line5_displayname:; Line 5 Display Name line5_authname: UNPROVISIONED ; Line 5 Registration Authentication line5_password: UNPROVISIONED ; Line 5 Registration Password # Line 6 Settings line6_name: ; Line 6 Extension\User ID line6_displayname:; Line 6 Display Name line6_authname: UNPROVISIONE ; Line 6 Registration Authentication line6_password: UNPROVISIONE ; Line 6 Registration Password # Emergency Proxy info proxy_emergency: proxy_emergency_port: 5060 # Backup Proxy info proxy_backup: proxy_backup_port: 5060 # Outbound Proxy info outbound_proxy: outbound_proxy_port: 5060 # NAT/Firewall Traversal nat_enable: 1 nat_address: 24.18.147.95 voip_control_port: 5060 start_media_port: 16384 end_media_port: 32766 nat_received_processing: 1 # Phone Label (Text desired to be displayed in upper right corner) phone_label: Ty's Phone ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: PST # Enable_VAD (1-enabled, 0-disabled) enable_vad: 0 # Network Media Type (auto, full100, full10, half100, half10) network_media_type: auto #user_info: phone # SIP Configuration File (stop) When the phone tries to register, all I get in the Asterisk console is this: Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804 handle_request_register: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '24.18.147.95' ...but the phone can make a call to any destination in the dialplan... :^/ Where's my stupidity? Am I confused on all the names in the .cnf file? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Saturday, April 23, 2005 6:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 won't register as SIP device It can use DNS if the DNS servers are valid. Can you post your SIP.conf? Didi you configure the phone manually or did you use the cnf files? If you used cnf files can you post those also? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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
Re: [Asterisk-Users] if outgoing call fails with provider 1 then auto try provider 2
Rich- wouldn't Andrew K's solution work? That seems to make good sense. There are no real examples that would address your points. The primary reason is that your * can dispatch a call to a provider and the provider will accept that handshaking call. But, if they are having internal call-completion issues, there is no way for you to know that. You could get some sort of busy, dead air, etc. You could probably design some sort of timer-based timeout, but what indication would you use to indicate the call was successful vs unsuccessful? There are several ways to address whether your * is successful in reaching your provider's equipment, but that's about it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] if outgoing call fails with provider 1 then auto try provider 2
Thanks Andrew for the great example! Anybody else have any input? Tom --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: On April 22, 2005 10:38 pm, Thomas Miller wrote: When someone teminates a call with my softphone to m __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Cisco 7960 won't register as SIP device
SNIP #user_info: phone # SIP Configuration File (stop) When the phone tries to register, all I get in the Asterisk console is this: Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804 handle_request_register: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '24.18.147.95' I am unfamiliar with the Cisco configs but I am just comparing your error message to what you have in the config to make this suggestion. In the error it has user=phone and in your config commented out there is #user_info: phone. What if you tried uncommenting that line and putting in username? It could be that when thatline is commented out, it uses phone by default. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.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: Hotel billing in IPSwitchBoard
Hi, Have a look at http://www.voip-info.org/wiki-CallingCard+Applications I recently used this in a hospital for the same concept. Can charge on caller ID etc. Works really well. Ties to a MySQL database, so a PHP interface can be coded to view the call charges etc on a room. It works on a card system, but all the SQL commands are customisable, so it does the job. Also, the destination charges are managable through the tables and different charges for different prefixes can be a applied. Also it supports LCDial (least cost routing dialler). So it will choose the carrier (if you box will use it) based on the cheapest rate (for the hotel, still charges the customer the same). In the application I used it for, it puts International Calls through our IP Provider and local calls/mobiles through our carrier as it was cheaper. Hope this might help, Thanks Mathew From: [EMAIL PROTECTED] on behalf of Chris Mason (Lists) Sent: Sat 23/04/2005 23:03 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard Also needed is a way to title and logo the print out so it looks like an invoice. A tempplate would work, and if can use HTML templates that would be easy to customise. Consider making the data a table that is substituted into the html template. Chris Mason www.anguillaguide.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of tgj Sent: Saturday, April 23, 2005 7:55 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard Exactly what I am looking for also. Because we have multiple phones in one villa, I would need the ability to group extensions and produce an overall bill, and I would, of course, need the ability to set the charge rate versus the cost, i.e., the cost is $.02/min, but we might charge $.50/min regardless of destination, a flat fee for all long distance and international. This is so cool. Hi Chris Grouping is a good idea, will not be in the first release, but later. There will only be a charge rate in the first release. You can charge depending on the destination. Thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.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 winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hotel billing in IPSwitchBoard
- Original Message - From: Thorben Jensen [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 8:11 AM Subject: [Asterisk-Users] Hotel billing in IPSwitchBoard I am currently working on implementing Hotel Billing in IPSwitchBoard. The idea is that a receptionist in a hotel can just right click an extension button and choose Account; IPS will now calculate the call charges made from that extension and show all calls and charges on a form. The receptionist now has the option to close the account which will reset the account. I will add a table for editing call charges, and there will be a possibility to add a fee for connection charges and also an option to charge calls per xx seconds and to add/subtract a percentage to all calls. I will add a family/key to the asterisk database to indicate if the extension is closed, this way you can stop outgoing calls from being made from a closed extension by checking the dial plan. Please let me know if there are any other features you would like to see in IPSwitchBoard. Hi, As mentioned before, how about being able to search and replay recordings from the switchboard. With call records now searchable hopefully it wouldn't take too much more work to enable. For example, being able to search on extension by date and time or by cli would be very handy. Best regards, Steve. ___ Asterisk-Users mailing list Asterisk-Users@lists.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 7960 won't register as SIP device
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb Sent: Saturday, April 23, 2005 11:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; List Receiver Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device SNIP #user_info: phone # SIP Configuration File (stop) When the phone tries to register, all I get in the Asterisk console is this: Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804 handle_request_register: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '24.18.147.95' I am unfamiliar with the Cisco configs but I am just comparing your error message to what you have in the config to make this suggestion. In the error it has user=phone and in your config commented out there is #user_info: phone. What if you tried uncommenting that line and putting in username? It could be that when thatline is commented out, it uses phone by default. Robert Actually after getting into the Cisco site it looks like you want a value of none for that. Configures the user= parameter in the REGISTER message. Valid values are: * none-No value is inserted. * phone-The value user=phone is inserted in the To, From, and Contact Headers for REGISTER. * ip-The value user=ip is inserted in the To, From, and Contact Headers for REGISTER. The default value is none. It says the default value is none but you may want to hard code it as it looks like that is not what it is doing. ___ Asterisk-Users mailing list Asterisk-Users@lists.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 7960 won't register as SIP device
Aye...that was it... Thanks a billion! -Original Message- From: Robert Webb [mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb Sent: Saturday, April 23, 2005 8:54 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion; List Receiver Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb Sent: Saturday, April 23, 2005 11:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; List Receiver Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device SNIP #user_info: phone # SIP Configuration File (stop) When the phone tries to register, all I get in the Asterisk console is this: Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804 handle_request_register: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '24.18.147.95' I am unfamiliar with the Cisco configs but I am just comparing your error message to what you have in the config to make this suggestion. In the error it has user=phone and in your config commented out there is #user_info: phone. What if you tried uncommenting that line and putting in username? It could be that when thatline is commented out, it uses phone by default. Robert Actually after getting into the Cisco site it looks like you want a value of none for that. Configures the user= parameter in the REGISTER message. Valid values are: * none-No value is inserted. * phone-The value user=phone is inserted in the To, From, and Contact Headers for REGISTER. * ip-The value user=ip is inserted in the To, From, and Contact Headers for REGISTER. The default value is none. It says the default value is none but you may want to hard code it as it looks like that is not what it is doing. 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
Re: [Asterisk-Users] Quadbri bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ?
Works for me too. We have an old fax machine sitting on the same NT1 as asterisk. In asterisk I ignored the MNS by setting the line exten = my_fax_msn,1,wait(30) Doesn't it work without the wait() in .nl? I just didn't mention the fax MSNs in my incoming context... I tried, but my default context only has a line: exten = s,1,Congestion I did that to prevent usage from outside, since my asterisk box is open for outside sip phones. My folks connect to it etc. So without the wait, the incoming call will search for an exten= line in the incoming context, won't find one so falls back to default,s,1 That way faxes wont arrive on my fax machine cause asterisk is playing the congestion tone. -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.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: Hotel billing in IPSwitchBoard
Hi, As mentioned before, how about being able to search and replay recordings from the switchboard. With call records now searchable hopefully it wouldn't take too much more work to enable. For example, being able to search on extension by date and time or by cli would be very handy. Best regards, Steve. Hi Steve, I will implement that too, but in a later release. thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Hotel billing in IPSwitchBoard
Now that makes me very excited. I have implemented a pbx in a datacenter for a online stock exchange and they want all calls recorded. I am uncertain how to handle recovery of the calls, though. This would be wonderful. Chris Mason www.anguillaguide.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Rawlings Sent: Saturday, April 23, 2005 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Hotel billing in IPSwitchBoard - Original Message - From: Thorben Jensen [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 8:11 AM Subject: [Asterisk-Users] Hotel billing in IPSwitchBoard I am currently working on implementing Hotel Billing in IPSwitchBoard. The idea is that a receptionist in a hotel can just right click an extension button and choose Account; IPS will now calculate the call charges made from that extension and show all calls and charges on a form. The receptionist now has the option to close the account which will reset the account. I will add a table for editing call charges, and there will be a possibility to add a fee for connection charges and also an option to charge calls per xx seconds and to add/subtract a percentage to all calls. I will add a family/key to the asterisk database to indicate if the extension is closed, this way you can stop outgoing calls from being made from a closed extension by checking the dial plan. Please let me know if there are any other features you would like to see in IPSwitchBoard. Hi, As mentioned before, how about being able to search and replay recordings from the switchboard. With call records now searchable hopefully it wouldn't take too much more work to enable. For example, being able to search on extension by date and time or by cli would be very handy. Best regards, Steve. ___ Asterisk-Users mailing list Asterisk-Users@lists.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
[Fwd: FW: [Asterisk-Users] IAX help]
Peter thanks for the response.
I put the user name and password in but i still get the same error.
;Extentions at telx-nyc
exten = _70XX,1,Dial(IAX2/telx-nyc:[EMAIL PROTECTED]/${EXTEN})
Apr 23 12:30:35 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected
connect attempt from 192.168.0.251
What else could it be?
-Original Message-
From: Peter Bowyer [mailto:[EMAIL PROTECTED]
Sent: Saturday, April 23, 2005 4:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX help
On 23/04/05, Michael DiMartino [EMAIL PROTECTED] wrote:
3. Extensions.conf (telx-NY17S)
;Extentions at telx-nyc
exten = _7XXX,1,Dial(IAX2/telx-nyc/${EXTEN})
exten = _7XXX,1,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN})
where username:password is the credientials you need to authenticate
with the other server.
The username/secret in iax2.conf is for inbound, not for outbound calls.
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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[Asterisk-Users] Re: Re: Hotel billing in IPSwitchBoard
Also needed is a way to title and logo the print out so it looks like an invoice. A tempplate would work, and if can use HTML templates that would be easy to customise. Consider making the data a table that is substituted into the html template. Chris Mason www.anguillaguide.com Hi Chris, I will find a solution :-) thank you thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.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] OctoBRI and 2.6kernel
are you using udev ? If yes, check README.udev in the zaptel directory On 4/23/05, Terry Wade [EMAIL PROTECTED] wrote: Hi Guys I am trying to get the Junghanns card to load on Suse 9.3 and tried to get it running on Fedora Core 3 (latest kernels). I have heard from a source here in South Africa that this is about as hard as pulling teeth. Could someone please confirm this for me and if they do have it working properly is it possible to get a guide. I can get the zaptel and qozap to load the card and all the ports and inside asterisk I see the zap channels. But I cannot get a line out or make any incoming calls. Are there some 2.6 tweaks that I need to do in the kernel. Kind Regards Terry Wade Mobile: +27 82 802-5750 Office: +27 11 784-7642 Fax: +27 11 388-0855 Linux is like a Wigwam - No gates, no windows, Apache inside Disclaimer and Confidentiality Warning This message is intended for the addressee only. If you are not the intended recipient of this message, you are notified that any distribution, use of or copying of this communication is strictly prohibited. If you have received the communication in error, please notify the sender immediately. The views and opinions expressed in this message are those of the individual sender of this message and do not necessarily represent the views and opinions of ActiCom. Consequently, ActiCom does not accept responsibility for such views and opinions and this message should not be read as representing the views and opinions of ActiCom without subsequent written confirmation. Each page attached hereto must also be read in conjunction with this disclaimer. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michal Bielicki http://www.aefirion.org/ http://www.asterisk.com.pl/ ___ Asterisk-Users mailing list Asterisk-Users@lists.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: [Fwd: FW: [Asterisk-Users] IAX help]
On 23/04/05, Michael DiMartino [EMAIL PROTECTED] wrote:
Peter thanks for the response.
I put the user name and password in but i still get the same error.
;Extentions at telx-nyc
exten = _70XX,1,Dial(IAX2/telx-nyc:[EMAIL PROTECTED]/${EXTEN})
Apr 23 12:30:35 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected
connect attempt from 192.168.0.251
What else could it be?
This peer entry in telx-nyc's iax.conf:
; telx-NY17S - Incoming
[telx-NY17S]
type=peer
secret=telx-NY17S
context=from-telx-NY17S
disallow=all
allow=ulaw
Needs to match with the dial string you're calling it with above. See
the difference?
Check the presented username with iax debug enabled to confirm.
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard
Taking this idea a little further. (I apreciate there may be legal issues with this request) Would it be possible for extensions to be tagged, so that if they make and / or recive a call the call is automatically recorded each and every time, at the end of the call the file is closed I would imagine, that its either set in the context menu of the extention (ie right click, select always record on active) or in the extensions list. A supervise (either on demand or always) would be a great help as well. On 4/23/05, tgj [EMAIL PROTECTED] wrote: Hi, As mentioned before, how about being able to search and replay recordings from the switchboard. With call records now searchable hopefully it wouldn't take too much more work to enable. For example, being able to search on extension by date and time or by cli would be very handy. Best regards, Steve. Hi Steve, I will implement that too, but in a later release. thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.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: [Fwd: FW: [Asterisk-Users] IAX help]
Peter Bowyer wrote:
On 23/04/05, Michael DiMartino [EMAIL PROTECTED] wrote:
Peter thanks for the response.
I put the user name and password in but i still get the same error.
;Extentions at telx-nyc
exten = _70XX,1,Dial(IAX2/telx-nyc:[EMAIL PROTECTED]/${EXTEN})
Apr 23 12:30:35 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected
connect attempt from 192.168.0.251
What else could it be?
This peer entry in telx-nyc's iax.conf:
; telx-NY17S - Incoming
[telx-NY17S]
type=peer
secret=telx-NY17S
context=from-telx-NY17S
disallow=all
allow=ulaw
Needs to match with the dial string you're calling it with above. See
the difference?
Check the presented username with iax debug enabled to confirm.
Peter
Peter, again thanks so much for your response. But not what u mean here.
i change the dial sting to following and i got same results.
;Extentions at telx-nyc
exten = _70XX,1,Dial(IAX2/telx-NY17S:[EMAIL PROTECTED]/${EXTEN})
What point am i missing?
Thanks in advance.
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Re: [Fwd: FW: [Asterisk-Users] IAX help]
Peter Bowyer wrote:
On 23/04/05, Michael DiMartino [EMAIL PROTECTED] wrote:
Peter thanks for the response.
I put the user name and password in but i still get the same error.
;Extentions at telx-nyc
exten = _70XX,1,Dial(IAX2/telx-nyc:[EMAIL PROTECTED]/${EXTEN})
Apr 23 12:30:35 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected
connect attempt from 192.168.0.251
What else could it be?
This peer entry in telx-nyc's iax.conf:
; telx-NY17S - Incoming
[telx-NY17S]
type=peer
secret=telx-NY17S
context=from-telx-NY17S
disallow=all
allow=ulaw
Needs to match with the dial string you're calling it with above. See
the difference?
Check the presented username with iax debug enabled to confirm.
Peter
Here is the output for iax2 debug on the telx-nyc server. the receicving
server
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 2ms SCall: 3 DCall: 0 [192.168.0.251:4569]
VERSION : 2
CALLED NUMBER : 7001
Unknown IE 045 : Present
CALLING NUMBER : 7101
Unknown IE 038 : Present
Unknown IE 039 : Present
Unknown IE 040 : Present
CALLING NAME: Telx 7101
LANGUAGE: en
USERNAME: telx-NY17S
FORMAT : 4
CAPABILITY : 2097151
ADSICPE : 2
DATE TIME : 177695713
Ignoring unknown information element 'Unknown IE' (45) of length 0
Ignoring unknown information element 'Unknown IE' (38) of length 1
Ignoring unknown information element 'Unknown IE' (39) of length 1
Ignoring unknown information element 'Unknown IE' (40) of length 2
Apr 23 13:34:01 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected
connect attempt from 192.168.0.251
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REJECT
Timestamp: 00016ms SCall: 3 DCall: 3 [192.168.0.251:4569]
CAUSE : No authority found
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 00016ms SCall: 3 DCall: 3 [192.168.0.251:4569]
Asterisk-NY60H*CLI
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[Asterisk-Users] IP Phones and firewalls ...
Hello all, Here is our problem .. IP SIP phones remote .. They will connect to our IP PBX (Asterisk Server) without issue however no voice makes it when anyone answers a phone call made by one of these IP phones. So this means SIP is working but RTP is not, Here is what I currently have on the firewall (http://m0n0.ch/wall). Firewall Rules TCP/UDP * * 192.168.2.253 5060 NAT SIP Protocol UDP * * 192.168.2.253 4569 NAT IAX Protocol UDP * * 192.168.2.253 5036 NAT IAX Protocol UDP * * 192.168.2.253 1 - 2 NAT RTP UDP NAT Rules WAN TCP/UDP 5060 - 5099 192.168.2.253 5060 SIP Protocol WAN UDP 4569 192.168.2.253 4569 IAX2 Protocol WAN UDP 5036 192.168.2.253 5036 IAX Protocol WAN UDP 1 - 2 192.168.2.253 1 - 2 RTP UDP Range IP phones are Sipra 841's and work great when on the same subnet as the * server, this only becomes an issue when offnet and of course outside of the firewall. So I am stumped as to why this does not work .. I have logging turned on for all of the above and see no packets getting dropped .. Anyone there able to shead some light on this .. Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Best of the best of IP Phones
Is there a specific SIP or IAX phone that truly shines above the rest where it comes to happy compatibility with Asterisk? I guess Im talking about feature sets, like early-dial, off hook call announcing, conferencing, echo suppression, etc etc. I, like many others, bought a Budgetone for early testing, and need some new eye candy! OHCA is a feature that Id love to integrate, and it seems that not too many phones support it out of the box. Chris Coulthurst [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] voice pulse connect - no dtmf
so how do we get this fixed, its happing to my one and only DID as well... On 4/22/05, Me [EMAIL PROTECTED] wrote: I had the same problem with another provider whom I got no response from as usual.. We had 5 or 6 numbers that worked fine and one that just quit sending DTMF. - Original Message - From: Doug Harris To: [EMAIL PROTECTED] Digium. Com Sent: Friday, April 22, 2005 11:52 AM Subject: [Asterisk-Users] voice pulse connect - no dtmf Hi, I've got bunch of VP connect lines, and a day back two LA area numbers stop sending DTMF. They are IAX2. So, simply my customers can dial in, it hit my IVR but when they punch-in the number, my * running 1.0.7 cannot identify the dtmf. IAX debug does not show dtmf being sent to me. Just want to know whether any of you had this experience, and if so how that was fixed. Funny thing is this happened on two dids and others are OK. Cheers DH ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Connecting Elmeg CS100 ISDN system phones to Asterisk
Hi, I still have a reasonable number of Elmeg CS100 ISDN system phones and also some Elmeg ISDN Dect sets lying around, which I ultimately would like to connect to my Asterisk system. Using the basic ISDN functionality using an NT capable ISDN card is no problem. I would however like to use the 'proprietary' Elmeg functions. The Elmeg phones are able to communicate with an Elmeg PABX for it's phonebook and other functions. It would be nice to emulate that in Asterisk, to make them more featurerich. Has anyone else ever tried to reverse-engineer these Elmeg extensions, or does anyone have a document describing it ? Or is there possibly already some software for Asterisk ? Thanks, Taco Scargo ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Most affordable 8-port NT-capable ISDN card
Hello, Does anyone know what the most affordable 8-port NT-capable ISDN card is (that is compatible with Asterisk) ? Thanks, Taco Scargo ___ Asterisk-Users mailing list Asterisk-Users@lists.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] voice pulse connect - no dtmf
Ours just started working again.. - Original Message - From: Justin Richards [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 1:14 AM Subject: Re: [Asterisk-Users] voice pulse connect - no dtmf so how do we get this fixed, its happing to my one and only DID as well... On 4/22/05, Me [EMAIL PROTECTED] wrote: I had the same problem with another provider whom I got no response from as usual.. We had 5 or 6 numbers that worked fine and one that just quit sending DTMF. - Original Message - From: Doug Harris To: [EMAIL PROTECTED] Digium. Com Sent: Friday, April 22, 2005 11:52 AM Subject: [Asterisk-Users] voice pulse connect - no dtmf Hi, I've got bunch of VP connect lines, and a day back two LA area numbers stop sending DTMF. They are IAX2. So, simply my customers can dial in, it hit my IVR but when they punch-in the number, my * running 1.0.7 cannot identify the dtmf. IAX debug does not show dtmf being sent to me. Just want to know whether any of you had this experience, and if so how that was fixed. Funny thing is this happened on two dids and others are OK. Cheers DH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Fwd: FW: [Asterisk-Users] IAX help]
On April 23, 2005 12:31 pm, Michael DiMartino wrote:
exten = _70XX,1,Dial(IAX2/telx-nyc:[EMAIL PROTECTED]/${EXTEN})
Apr 23 12:30:35 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected
connect attempt from 192.168.0.251
The extension you're hitting doesn't exist in the context you are being dumped
in to telx-nyc [telx-nyc] type=user entry?
The codec you are wanting and what they are offering don't match?
Bad password?
turn on iax debugging and see if you get more detail.
-A.
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Re: [Fwd: FW: [Asterisk-Users] IAX help]
Peter Bowyer wrote:
On 23/04/05, Michael DiMartino [EMAIL PROTECTED] wrote:
Peter thanks for the response.
I put the user name and password in but i still get the same error.
;Extentions at telx-nyc
exten = _70XX,1,Dial(IAX2/telx-nyc:[EMAIL PROTECTED]/${EXTEN})
Apr 23 12:30:35 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected
connect attempt from 192.168.0.251
What else could it be?
This peer entry in telx-nyc's iax.conf:
; telx-NY17S - Incoming
[telx-NY17S]
type=peer
secret=telx-NY17S
context=from-telx-NY17S
disallow=all
allow=ulaw
Needs to match with the dial string you're calling it with above. See
the difference?
Check the presented username with iax debug enabled to confirm.
Peter
this is the output for iax2 show users
Asterisk-NY60H*CLI iax2 show users
Username SecretAuthen Def.Context
A/C
stealth telxvoip 002 from-jnctn
Yes
telx-atl telx-atl 003 from-telx-atl
No
jnctnKey: jnctn004 from-jnctn
No
guest-no secret- 003 default
No
Asterisk-NY60H*CLI
the the telx-NY17S is not their. Strange.
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Re: [Fwd: FW: [Asterisk-Users] IAX help]
On April 23, 2005 12:39 pm, Peter Bowyer wrote: ; telx-NY17S - Incoming [telx-NY17S] type=peer secret=telx-NY17S context=from-telx-NY17S disallow=all allow=ulaw I think your nomenclature's wrong. when I call an IAX host, I look for a matching type=peer entry in my iax.conf. When I receive a call from an IAX host, I look for a matching type=user entry in my iax.conf. putting a context= in a peer entry doesn't do anything truly useful, since the far side's type=user entry will determine the context unless I manually specify it in the dial string. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard
(I apreciate there may be legal issues with this request) The only legal issue is, I believe, you have too announce to the caller This call may be recorded etc... You are entitled to use call recording, if in doubt put a big sign up in the room Calls are recorded I think it's not an issue we have to worry about here. 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
[Asterisk-Users] Dialing problem - Cisco 7290 to anything
Hi All,
Still having problems :-(
I have an Asterisk 1-0-7 setup on Debian 3.1 (Sparc)
I have severel SIP phones that call between each other and can chat no
probs. I can even call from the SIP phones to the sccp 7920 no
probs
However when I call from the 7290 to any SIP phone it just doesn't recognise
that the other person has answered the SIP phone, it just carries on making
the 'ringing' noise. When I hit hangup, the display of the 7290 changes to
onhook state but I can still hear the ringing
Any Ideas?
here are some copies of my config..
sccp.conf
[general]
keepalive = 5
context = home
dateFormat = D-M-Y ; M-D-Y in any order (5 chars max)
bindaddr = 192.122.122.22;
port = 2000; listen on port 2000 (Skinny, default)
[SEP000D282E89AA]
description = Walnuts Wireless
type = 7920
context = home
tzoffset = 0
autologin = wireless
[wireless]
id = 2210
context = home
callwaiting = 1
mailbox = 2210
callerid= Wireless, 2210
extensions.conf
[globals]
PHONES10=SCCP/wireless
PHONES10VM=wireless
[home]
exten = 2210,1,SetCalledParty(wireless 2000)
exten = 2000,2,Dial(SCCP/wireless)
exten = 2210,3,Macro(vmessage,${PHONES10VM})
exten = 2210,4,Hangup
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[Asterisk-Users] ASTERISK PROGRAMER
5 Settings line5_name: ; Line 5 Extension\User ID line5_displayname:; Line 5 Display Name line5_authname: UNPROVISIONED ; Line 5 Registration Authentication line5_password: UNPROVISIONED ; Line 5 Registration Password # Line 6 Settings line6_name: ; Line 6 Extension\User ID line6_displayname:; Line 6 Display Name line6_authname: UNPROVISIONE ; Line 6 Registration Authentication line6_password: UNPROVISIONE ; Line 6 Registration Password # Emergency Proxy info proxy_emergency: proxy_emergency_port: 5060 # Backup Proxy info proxy_backup: proxy_backup_port: 5060 # Outbound Proxy info outbound_proxy: outbound_proxy_port: 5060 # NAT/Firewall Traversal nat_enable: 1 nat_address: 24.18.147.95 voip_control_port: 5060 start_media_port: 16384 end_media_port: 32766 nat_received_processing: 1 # Phone Label (Text desired to be displayed in upper right corner) phone_label: Ty's Phone ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: PST # Enable_VAD (1-enabled, 0-disabled) enable_vad: 0 # Network Media Type (auto, full100, full10, half100, half10) network_media_type: auto #user_info: phone # SIP Configuration File (stop) When the phone tries to register, all I get in the Asterisk console is this: Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804 handle_request_register: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '24.18.147.95' ...but the phone can make a call to any destination in the dialplan... :^/ Where's my stupidity? Am I confused on all the names in the .cnf file? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Saturday, April 23, 2005 6:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 won't register as SIP device It can use DNS if the DNS servers are valid. Can you post your SIP.conf? Didi you configure the phone manually or did you use the cnf files? If you used cnf files can you post those also? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- next part -- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3032 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050423/3b f4397b/smime-0001.bin -- Message: 2 Date: Sat, 23 Apr 2005 08:25:29 -0700 (PDT) From: Thomas Miller [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] if outgoing call fails with provider 1 then auto try provider 2 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Rich- wouldn't Andrew K's solution work? That seems to make good sense. There are no real examples that would address your points. The primary reason is that your * can dispatch a call to a provider and the provider will accept that handshaking call. But, if they are having internal call-completion issues, there is no way for you to know that. You could get some sort of busy, dead air, etc. You could probably design some sort of timer-based timeout, but what indication would you use to indicate the call was successful vs unsuccessful? There are several ways to address whether your * is successful in reaching your provider's equipment, but that's about it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -- Message: 3 Date: Sat, 23 Apr 2005 08:26:25 -0700 (PDT) From: Thomas Miller [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] if outgoing call fails with provider 1 then auto try provider 2 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Thanks Andrew for the great example! Anybody else have any input? Tom --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: On April 22, 2005 10:38 pm, Thomas Miller wrote: When someone teminates a call with my softphone to m __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection
[Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local
I'm setting up asterisk and want everything to load on startup. The distro I'm using is a RHEL4 rebuild (CentOS4). Because the zaptel init script doesn't work I'm trying to set everything up from rc.local. However asterisk fails to start with an error that the zap device isn't loaded. This is what I do in rc.local: /sbin/modprobe wcfxs sleep 2 /sbin/ztcfg - sleep 5 /etc/rc.d/init.d/asterisk start Zaptel does display that it has configured the 2 channels but in fact, it didn't. When I re-run ztcfg then asterisk does start. Why does it do this? 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] ASTERISK PROGRAMER
Registration Password # Line 4 Settings line4_name: ; Line 4 Extension\User ID line4_displayname:; Line 4 Display Name line4_authname: UNPROVISIONED ; Line 4 Registration Authentication line4_password: UNPROVISIONED ; Line 4 Registration Password # Line 5 Settings line5_name: ; Line 5 Extension\User ID line5_displayname:; Line 5 Display Name line5_authname: UNPROVISIONED ; Line 5 Registration Authentication line5_password: UNPROVISIONED ; Line 5 Registration Password # Line 6 Settings line6_name: ; Line 6 Extension\User ID line6_displayname:; Line 6 Display Name line6_authname: UNPROVISIONE ; Line 6 Registration Authentication line6_password: UNPROVISIONE ; Line 6 Registration Password # Emergency Proxy info proxy_emergency: proxy_emergency_port: 5060 # Backup Proxy info proxy_backup: proxy_backup_port: 5060 # Outbound Proxy info outbound_proxy: outbound_proxy_port: 5060 # NAT/Firewall Traversal nat_enable: 1 nat_address: 24.18.147.95 voip_control_port: 5060 start_media_port: 16384 end_media_port: 32766 nat_received_processing: 1 # Phone Label (Text desired to be displayed in upper right corner) phone_label: Ty's Phone ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: PST # Enable_VAD (1-enabled, 0-disabled) enable_vad: 0 # Network Media Type (auto, full100, full10, half100, half10) network_media_type: auto #user_info: phone # SIP Configuration File (stop) When the phone tries to register, all I get in the Asterisk console is this: Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804 handle_request_register: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '24.18.147.95' ...but the phone can make a call to any destination in the dialplan... :^/ Where's my stupidity? Am I confused on all the names in the .cnf file? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Saturday, April 23, 2005 6:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 won't register as SIP device It can use DNS if the DNS servers are valid. Can you post your SIP.conf? Didi you configure the phone manually or did you use the cnf files? If you used cnf files can you post those also? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- next part -- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3032 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050423/3b f4397b/smime-0001.bin -- Message: 2 Date: Sat, 23 Apr 2005 08:25:29 -0700 (PDT) From: Thomas Miller [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] if outgoing call fails with provider 1 then auto try provider 2 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Rich- wouldn't Andrew K's solution work? That seems to make good sense. There are no real examples that would address your points. The primary reason is that your * can dispatch a call to a provider and the provider will accept that handshaking call. But, if they are having internal call-completion issues, there is no way for you to know that. You could get some sort of busy, dead air, etc. You could probably design some sort of timer-based timeout, but what indication would you use to indicate the call was successful vs unsuccessful? There are several ways to address whether your * is successful in reaching your provider's equipment, but that's about it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -- Message: 3 Date: Sat, 23 Apr 2005 08:26:25 -0700 (PDT) From: Thomas Miller [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] if outgoing call fails with provider 1 then auto try provider 2 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Thanks Andrew for the great example! Anybody else
Re: [Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local
You need this before wcfxs /sbin/modprobe zaptel Regards, Chris - Original Message - From: Remco Barende [EMAIL PROTECTED] To: Asterisk Users List asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 12:55 PM Subject: [Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local I'm setting up asterisk and want everything to load on startup. The distro I'm using is a RHEL4 rebuild (CentOS4). Because the zaptel init script doesn't work I'm trying to set everything up from rc.local. However asterisk fails to start with an error that the zap device isn't loaded. This is what I do in rc.local: /sbin/modprobe wcfxs sleep 2 /sbin/ztcfg - sleep 5 /etc/rc.d/init.d/asterisk start Zaptel does display that it has configured the 2 channels but in fact, it didn't. When I re-run ztcfg then asterisk does start. Why does it do this? 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] ASTERISK PROGRAMER
; Line 4 Registration Password # Line 5 Settings line5_name: ; Line 5 Extension\User ID line5_displayname:; Line 5 Display Name line5_authname: UNPROVISIONED ; Line 5 Registration Authentication line5_password: UNPROVISIONED ; Line 5 Registration Password # Line 6 Settings line6_name: ; Line 6 Extension\User ID line6_displayname:; Line 6 Display Name line6_authname: UNPROVISIONE ; Line 6 Registration Authentication line6_password: UNPROVISIONE ; Line 6 Registration Password # Emergency Proxy info proxy_emergency: proxy_emergency_port: 5060 # Backup Proxy info proxy_backup: proxy_backup_port: 5060 # Outbound Proxy info outbound_proxy: outbound_proxy_port: 5060 # NAT/Firewall Traversal nat_enable: 1 nat_address: 24.18.147.95 voip_control_port: 5060 start_media_port: 16384 end_media_port: 32766 nat_received_processing: 1 # Phone Label (Text desired to be displayed in upper right corner) phone_label: Ty's Phone ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: PST # Enable_VAD (1-enabled, 0-disabled) enable_vad: 0 # Network Media Type (auto, full100, full10, half100, half10) network_media_type: auto #user_info: phone # SIP Configuration File (stop) When the phone tries to register, all I get in the Asterisk console is this: Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804 handle_request_register: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '24.18.147.95' ...but the phone can make a call to any destination in the dialplan... :^/ Where's my stupidity? Am I confused on all the names in the .cnf file? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Saturday, April 23, 2005 6:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 won't register as SIP device It can use DNS if the DNS servers are valid. Can you post your SIP.conf? Didi you configure the phone manually or did you use the cnf files? If you used cnf files can you post those also? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- next part -- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3032 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050423/3b f4397b/smime-0001.bin -- Message: 2 Date: Sat, 23 Apr 2005 08:25:29 -0700 (PDT) From: Thomas Miller [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] if outgoing call fails with provider 1 then auto try provider 2 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Rich- wouldn't Andrew K's solution work? That seems to make good sense. There are no real examples that would address your points. The primary reason is that your * can dispatch a call to a provider and the provider will accept that handshaking call. But, if they are having internal call-completion issues, there is no way for you to know that. You could get some sort of busy, dead air, etc. You could probably design some sort of timer-based timeout, but what indication would you use to indicate the call was successful vs unsuccessful? There are several ways to address whether your * is successful in reaching your provider's equipment, but that's about it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -- Message: 3 Date: Sat, 23 Apr 2005 08:26:25 -0700 (PDT) From: Thomas Miller [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] if outgoing call fails with provider 1 then auto try provider 2 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Thanks Andrew for the great example! Anybody else have any input? Tom --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: On April 22, 2005 10:38 pm, Thomas Miller wrote: When someone teminates a call with my softphone to m __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Re: [Asterisk-Users] ASTERISK PROGRAMER
Authentication line3_password: UNPROVISIONED ; Line 3 Registration Password # Line 4 Settings line4_name: ; Line 4 Extension\User ID line4_displayname:; Line 4 Display Name line4_authname: UNPROVISIONED ; Line 4 Registration Authentication line4_password: UNPROVISIONED ; Line 4 Registration Password # Line 5 Settings line5_name: ; Line 5 Extension\User ID line5_displayname:; Line 5 Display Name line5_authname: UNPROVISIONED ; Line 5 Registration Authentication line5_password: UNPROVISIONED ; Line 5 Registration Password # Line 6 Settings line6_name: ; Line 6 Extension\User ID line6_displayname:; Line 6 Display Name line6_authname: UNPROVISIONE ; Line 6 Registration Authentication line6_password: UNPROVISIONE ; Line 6 Registration Password # Emergency Proxy info proxy_emergency: proxy_emergency_port: 5060 # Backup Proxy info proxy_backup: proxy_backup_port: 5060 # Outbound Proxy info outbound_proxy: outbound_proxy_port: 5060 # NAT/Firewall Traversal nat_enable: 1 nat_address: 24.18.147.95 voip_control_port: 5060 start_media_port: 16384 end_media_port: 32766 nat_received_processing: 1 # Phone Label (Text desired to be displayed in upper right corner) phone_label: Ty's Phone ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: PST # Enable_VAD (1-enabled, 0-disabled) enable_vad: 0 # Network Media Type (auto, full100, full10, half100, half10) network_media_type: auto #user_info: phone # SIP Configuration File (stop) When the phone tries to register, all I get in the Asterisk console is this: Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804 handle_request_register: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '24.18.147.95' ...but the phone can make a call to any destination in the dialplan... :^/ Where's my stupidity? Am I confused on all the names in the .cnf file? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Saturday, April 23, 2005 6:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 won't register as SIP device It can use DNS if the DNS servers are valid. Can you post your SIP.conf? Didi you configure the phone manually or did you use the cnf files? If you used cnf files can you post those also? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- next part -- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3032 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050423/3b f4397b/smime-0001.bin -- Message: 2 Date: Sat, 23 Apr 2005 08:25:29 -0700 (PDT) From: Thomas Miller [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] if outgoing call fails with provider 1 then auto try provider 2 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Rich- wouldn't Andrew K's solution work? That seems to make good sense. There are no real examples that would address your points. The primary reason is that your * can dispatch a call to a provider and the provider will accept that handshaking call. But, if they are having internal call-completion issues, there is no way for you to know that. You could get some sort of busy, dead air, etc. You could probably design some sort of timer-based timeout, but what indication would you use to indicate the call was successful vs unsuccessful? There are several ways to address whether your * is successful in reaching your provider's equipment, but that's about it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -- Message: 3 Date: Sat, 23 Apr 2005 08:26:25 -0700 (PDT) From: Thomas Miller [EMAIL PROTECTED] Subject: Re: [Asterisk-Users
RE: [Asterisk-Users] IP Phones and firewalls ...
It shows the public IP and not the private IP .. 403/403 67.181.191.99 D N 255.255.255.255 5061 Unmonitored BRW -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Saturday, April 23, 2005 6:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IP Phones and firewalls ... When you do a sip show peers from the what IP address does it show for the 841? - Original Message - From: Brian Watters [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 12:52 AM Subject: [Asterisk-Users] IP Phones and firewalls ... Hello all, Here is our problem .. IP SIP phones remote .. They will connect to our IP PBX (Asterisk Server) without issue however no voice makes it when anyone answers a phone call made by one of these IP phones. So this means SIP is working but RTP is not, Here is what I currently have on the firewall (http://m0n0.ch/wall). Firewall Rules TCP/UDP * * 192.168.2.253 5060 NAT SIP Protocol UDP * * 192.168.2.253 4569 NAT IAX Protocol UDP * * 192.168.2.253 5036 NAT IAX Protocol UDP * * 192.168.2.253 1 - 2 NAT RTP UDP NAT Rules WAN TCP/UDP 5060 - 5099 192.168.2.253 5060 SIP Protocol WAN UDP 4569 192.168.2.253 4569 IAX2 Protocol WAN UDP 5036 192.168.2.253 5036 IAX Protocol WAN UDP 1 - 2 192.168.2.253 1 - 2 RTP UDP Range IP phones are Sipra 841's and work great when on the same subnet as the * server, this only becomes an issue when offnet and of course outside of the firewall. So I am stumped as to why this does not work .. I have logging turned on for all of the above and see no packets getting dropped .. Anyone there able to shead some light on this .. Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best of the best of IP Phones
IMHO its the Cisco 7960. I have 5 of them littered around mu house. My wife uses the intercom feature to hunt me down when she has honey do lists for me. I must get around to breaking that feature ;-} Chris Coulthurst wrote: Is there a specific SIP or IAX phone that truly shines above the rest where it comes to happy compatibility with Asterisk? I guess Im talking about feature sets, like early-dial, off hook call announcing, conferencing, echo suppression, etc etc. I, like many others, bought a Budgetone for early testing, and need some new eye candy! OHCA is a feature that Id love to integrate, and it seems that not too many phones support it out of the box. **/Chris Coulthurst/** //[EMAIL PROTECTED]// mailto:[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
[Asterisk-Users] How to replace VM busy.gsm and unavail.gsm messages with custom files
Ive tried to replace the gsm, wav and WAV files in the /var/spool/asterisk/vm/default/201 directory with some strung-together Allison files, but every time I try, it just plays the default greet. Is this possible, or is it just that Im doing something wrong? Chris Coulthurst [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] Provisioning Lines
Hi, This may be a dumb question but I know how to provision lines but what is the use for them. Right now I just have one line provisioned on my cisco 7690 and I get all incoming calls on that line and make calls on that too. Additional lines may be mean additional extension numbers. But then why would a person want to have six different extensions and remember them all. One feature I could think off is 1.) To have a second line as auto-answer for paging, etc. Please dont flame me. Just getting into PBXs and havent had much experience with them. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ASTERISK PROGRAMER
On Saturday 23 April 2005 19:23, Bob Goddard wrote: On Saturday 23 April 2005 19:13, Matt Klein wrote: $4,172.38 USD and I'll programin anything you want for asterisk server. You are too stupid for the job. Quoting the 1200-line long Asterisk Digest message in your reply and adding one single line to it, where you just insult someone who was making a joke and add nothing of value is also stupid. People who live in glass houses shouldn't throw stones... Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ASTERISK PROGRAMER
The funniest part is, he thought I was serious. I'd be dumb if I didn't at least charge $4,172.39 USD for the job. On Sat, 23 Apr 2005, Gary Stimson wrote: On Saturday 23 April 2005 19:23, Bob Goddard wrote: On Saturday 23 April 2005 19:13, Matt Klein wrote: $4,172.38 USD and I'll programin anything you want for asterisk server. You are too stupid for the job. Quoting the 1200-line long Asterisk Digest message in your reply and adding one single line to it, where you just insult someone who was making a joke and add nothing of value is also stupid. People who live in glass houses shouldn't throw stones... Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Best of the best of IP Phones
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have to agree, the Cisco 7960 is probably the best (I have yet to try a 7970/71). Cisco are a pain to deal with (they only want to deal with large value customers/distributors) and the phone do have some small quirks/bugs but they are the best in functionality and build quality. They are also the best speaker phone for small conferences. - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD:519961 Gossiptel:9309811 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBQmqolktP/KMNOfRbAQKunAgAjDNbdUk9pf1psj9HeRFOila9Je/5G+fk +9BCLiaFX6dHwlMypkP1kCSSn09GIHgPlOW2TjQnixKWai20m7H7Kg9TyhppHsO/ Ehzz3oZonDalqjZnb7FhcfW2pvokBaNTGPtABGjNsffx6d4AO5X63VZ505DoRM7W S1kZAeUeuAAsrr7a8L3lm7o0zRRRJvKUCkCk0u/tW9/LdbdhCd3Z+ISunb3KoIXQ k6fXwhP4O61iSZsHRo3m42UR+ha3HHn1ZL/jYOJ3IUJYexJLDnmoxidmW1xjis3o l92eteUHq48AHyHw2JtdvdDdfl9XDOyDAyStCGrwXIXPUprFH/QgBQ== =+bDS -END PGP 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: Hotel billing in IPSwitchBoard
Depends on the state you are in. In Nebraska there is no law saying you have to tell someone they are being recorded if you are recording them on a business line. In Iowa you don't have to tell them , but you have to play a tone in the background every so many seconds. - Original Message - From: Chris Mason (Lists) [EMAIL PROTECTED] To: 'David John Walsh' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 12:40 PM Subject: RE: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard (I apreciate there may be legal issues with this request) The only legal issue is, I believe, you have too announce to the caller This call may be recorded etc... You are entitled to use call recording, if in doubt put a big sign up in the room Calls are recorded I think it's not an issue we have to worry about here. 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Provisioning Lines
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Manjit Riat wrote: Hi, This may be a dumb question but I know how to provision lines but what is the use for them. Right now I just have one line provisioned on my cisco 7690 and I get all incoming calls on that line and make calls on that too. Additional lines may be mean additional extension numbers. But then why would a person want to have six different extensions and remember them all. One feature I could think off is 1.) To have a second line as auto-answer for paging, etc. Please don?t flame me. Just getting into PBX?s and haven?t had much experience with them. 2.) Registering the phone with more than one server (possibly in different parts of the world). 3.) Different caller IDs Personally, I use the extra line buttons as speed dials. - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD:519961 Gossiptel:9309811 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBQmqpn0tP/KMNOfRbAQIxLAgAsx6pyAHjOOBWY6I6B+THNF3wjju5BsMe sSQFAV5oacuFxIjt+ai03fhQKOeno8nnpOJ7udvpNzss6EUiZl3VRLfjUUiGZGX5 07+v9NeLWnjQM8lt4ndJh+BLbYgMBihgG61dJF0h2a5UIc3ms7s0Qm82YOz2+yXt 1zJh1BYZECTdTs9NHQCtBsDoxE8JWiaRN+pJRe5A38YDcnalsZf0JDK08dBxXwhh 3rXU56TTyJTKuwIsd7TCjG4zSdibdB52lC2y2wGBd869Xfekn4NpWH2KfsItyxWV KTbsFNlfE850OiqAOnwW32FH6kktPlNUMD4kpd3L9PFp1Xav64w8zA== =MVRZ -END PGP 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] How to replace VM busy.gsm and unavail.gsmmessages with custom files
Also on a side note, previously read documentation aside, is there a surefire way to concatenate .gsm files together without getting any errors or losing quality due to re-conversions? I took several .gsm files from the Allison pack and cated them together. When I tried to Playback(File) it didnt play. Just dead air. So, I thought Id try and sox convert it to a wav, then back to a GSM file, which spit out some errors, but made the file anyway. This NEW file does work, but it has lost some of the quality, adding a few pops. Chris Coulthurst [EMAIL PROTECTED] ---BeginMessage--- Ive tried to replace the gsm, wav and WAV files in the /var/spool/asterisk/vm/default/201 directory with some strung-together Allison files, but every time I try, it just plays the default greet. Is this possible, or is it just that Im doing something wrong? Chris Coulthurst [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---End 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] ignorepat doesn't work
Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the ignorepat in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. thanks again rgds bk --Apple-Mail-18--1172348 Content-Transfer-Encoding: 7bit Content-Type: text/enriched; charset=US-ASCII thanks On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: colorparam,,DEDE/paramI had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. /color thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. colorparam,,DEDE/param Not sure if this is a bug or a feature. /color probably intentional. colorparam,,DEDE/paramSo, try placing the ignorepat in your handset-contexts instead. /color Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. thanks again rgds bk --Apple-Mail-18--1172348-- ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ignorepat doesn't work
ignorepat is for Zapata devices. Sip devices sned the number to the swith AFTER the SIP device feels it has dialed it. I am not a pro on the GS phones, (never played with them) but I would cheak the documentation on setting up a 'dialplan'. I hope this sets you in the right direction. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaime Blanco Sent: Saturday, April 23, 2005 4:38 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the ignorepat in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. thanks again rgds bk --Apple-Mail-18--1172348 Content-Transfer-Encoding: 7bit Content-Type: text/enriched; charset=US-ASCII thanks On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: colorparam,,DEDE/paramI had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. /color thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. colorparam,,DEDE/param Not sure if this is a bug or a feature. /color probably intentional. colorparam,,DEDE/paramSo, try placing the ignorepat in your handset-contexts instead. /color Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. thanks again rgds bk --Apple-Mail-18--1172348-- ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Re: ztcfg doesn't do anything from /etc/rc.d/rc.local
In article [EMAIL PROTECTED], Remco Barende [EMAIL PROTECTED] wrote: Because the zaptel init script doesn't work I'm trying to set everything up from rc.local. You should instead find out why the zaptel init script doesn't work for you. It works fine for me under Fedora Core 3, with one exception: I needed to increase the number of loop iterations where it waits for /dev/zap to become available, by changing TMOUT=10 to TMOUT=20 in /etc/rc.d/init.d/zaptel (my system was taking 11 or 12 iterations). Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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
Re: [Asterisk-Users] Provisioning Lines
At 02:07 PM 4/23/2005, you wrote: Hi, This may be a dumb question but I know how to provision lines but what is the use for them. Right now I just have one line provisioned on my cisco 7690 and I get all incoming calls on that line and make calls on that too. Additional lines may be mean additional extension numbers. But then why would a person want to have six different extensions and remember them all. One feature I could think off is 1.) To have a second line as auto-answer for paging, etc. Please don't flame me. Just getting into PBX's and haven't had much experience with them. Same extension but using call waiting... So when your girlfriend calls while you are talking on the phone, you can see the incoming call and callerID, and then either ignore it and let it bounce to vmail or put your wife on hold and take the call from your girlfriend. It really depends on your relationships. Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Provisioning Lines
I don't know about other phones but on the Sipura, I set all the lines to the same extension and incoming calls rollover on to the next line appearance. Hence, I can hold one and take the next call,switch back and forth easily, works great. Anotherreason would be to have more than one incoming DID, to make sure they were answered even if you were on the phone, you would see that youhad a call on DID2 and wouold put thefirst on hold to answer the second. You might want tohave a internal only extension. Chris Mason www.anguillaguide.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Manjit RiatSent: Saturday, April 23, 2005 3:08 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Provisioning Lines Hi, This may be a dumb question but I know how to provision lines but what is the use for them. Right now I just have one line provisioned on my cisco 7690 and I get all incoming calls on that line and make calls on that too. Additional lines may be mean additional extension numbers. But then why would a person want to have six different extensions and remember them all. One feature I could think off is 1.) To have a second line as auto-answer for paging, etc. Please dont flame me. Just getting into PBXs and havent had much experience with them. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Best of the best of IP Phones
I just got a cisco 7960, a bit tough to get going at first but it's a great phone. Supports OHVA, and the dialplan is very nice in that you can have autocompletion based on your plan. for example, if I dial 300, the phone completes, whereas if i dial a 1+ number there is a timeout. For OHVA, you choose a line to use, and you can have calls autoanswer, so its not much OHVA but rather like a pbx autoanswer. It is an expensive phone, but to give you an idea I just replaced my $500 original pingtel phone with the $300 cisco 7960. Also, sound quality is excellent. All I wish it had is an actual hold button rather than using the screen. Greg From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris CoulthurstSent: Saturday, April 23, 2005 2:10 AMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] Best of the best of IP Phones Is there a specific SIP or IAX phone that truly shines above the rest where it comes to happy compatibility with Asterisk? I guess Im talking about feature sets, like early-dial, off hook call announcing, conferencing, echo suppression, etc etc. I, like many others, bought a Budgetone for early testing, and need some new eye candy! OHCA is a feature that Id love to integrate, and it seems that not too many phones support it out of the box. Chris Coulthurst [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] ignorepat doesn't work
Grandstream does not support a dialplan. It is supposed to support Early Dial, but didn't work. I've been told that recent firmware fixes the early dial bug. I doubt that Early Dial is the solution. The solution is to buy a good IP Phone. Polycom and SIPura both support continue dialtone after digit. Cisco ATAs do not. I don't know if the Cisco IP phones do or not. Alexander Lopez wrote: ignorepat is for Zapata devices. Sip devices sned the number to the swith AFTER the SIP device feels it has dialed it. I am not a pro on the GS phones, (never played with them) but I would cheak the documentation on setting up a 'dialplan'. I hope this sets you in the right direction. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaime Blanco Sent: Saturday, April 23, 2005 4:38 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the ignorepat in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. -- 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
Re: [Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local
Chris wrote: You need this before wcfxs /sbin/modprobe zaptel *sigh* zaptel will automatically load when the card driver loads. modporbe will also run ztcfg after loading the card driver because (if you ran make install) /etc/modules.conf tells it to do so. -- 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
Re: [Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local
*sigh* I always get an error if I don't. Regards, Chris - Original Message - From: Eric Wieling aka ManxPower [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 7:15 PM Subject: Re: [Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local Chris wrote: You need this before wcfxs /sbin/modprobe zaptel *sigh* zaptel will automatically load when the card driver loads. modporbe will also run ztcfg after loading the card driver because (if you ran make install) /etc/modules.conf tells it to do so. -- 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
RE: [Asterisk-Users] callto: URL (URI) tag for dialing
I just wrote a simple cgi to have a form generate the number, then the cgi creates a call file and bingo. Web call. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Elkins Sent: Friday, April 22, 2005 8:21 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] callto: URL (URI) tag for dialing I see that there seems to be a 'callto' URL/URI for dialling a phone number... ie - on my web site's Contact Page - I have added the code... a href=callto:+27128070590+27 12 807-0590/a There should be some generic way for Mozilla (firefox - etc) to somehow turn a click on such a link into persuading Asterisk to dial the number for me and connect it to my SIP hard-phone. 1 - mini application under mozilla to collect the number/sip address, add in a static local extension (personal settings?) and pass info to a listener (auto-dialer) on the Asterisk Machine 2 - Auto Dialer dials my extension, then on answer, dials the URL or phone number. The URL could either be a simple phone number or a full SIP address?? Anyone done this? ..and care to share? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ignorepat doesn't work
Try adding a comma to your digitmap where you wish the dialtone to come back on. Works on a Polycom. On Apr 23, 2005, at 7:12 PM, Eric Wieling aka ManxPower wrote: Grandstream does not support a dialplan. It is supposed to support Early Dial, but didn't work. I've been told that recent firmware fixes the early dial bug. I doubt that Early Dial is the solution. The solution is to buy a good IP Phone. Polycom and SIPura both support continue dialtone after digit. Cisco ATAs do not. I don't know if the Cisco IP phones do or not. Alexander Lopez wrote: ignorepat is for Zapata devices. Sip devices sned the number to the swith AFTER the SIP device feels it has dialed it. I am not a pro on the GS phones, (never played with them) but I would cheak the documentation on setting up a 'dialplan'. I hope this sets you in the right direction. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaime Blanco Sent: Saturday, April 23, 2005 4:38 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the ignorepat in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. -- 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] SIP registration behind Linksys WRT54G
Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri=sip:asterisk.mydomain.com, ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri=sip 66.x.x.166:5060, ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.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 registration behind Linksys WRT54G
Please make sure you post any solution you find to this issue to the list I have been frustrated by this as well. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri=sip:asterisk.mydomain.com, ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri=sip 66.x.x.166:5060, ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.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] SIP registration behind Linksys WRT54G
Is your problem on the same model of Linksys? WRT54G? I haven't had a chance to try some other Linksys routers so I'm curious. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Henderson Sent: Saturday, April 23, 2005 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Please make sure you post any solution you find to this issue to the list I have been frustrated by this as well. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri=sip:asterisk.mydomain.com, ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri=sip 66.x.x.166:5060, ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ADSI Input from 480 keypad?
Couldn't you just assign the OK softkey to send a '#'? Seems that my local telco (with my 390 phone) sends many macros that are just DTMF sent from the softkeys. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Jimmy |Sent: Monday, April 11, 2005 10:43 AM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] ADSI Input from 480 keypad? | |Sorry it this is a reposting, but I got no replies last time, and wanted to |make sure that the post went through. | |Setup: |Asterisk 1.0.5 |Aastra 480 analog ADSI phone | | |Is it possible to program a Aastra 480 phone to accept input from the |keypad in response to a screen prompt? | |For example, the screen says | |Enter Password |And Press Ok | |I can make the text appear, I can create a softkey that says Ok, I |can get a prompt for the input, and the text shows up when I type on the |keypad. But how do I end the input stream, and send the input on it's |way? (i.e. press the enter key on a computer) | |Thanks for any 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] TE11OP - Mitel 200Sx??
The Switch is since 1995 and I get a SX-200 Digital G1005 ENH 672P / F25.0
09-FEB1994 when I look up the software on the switch board so if I am
reading what your telling me then I have to do D4/AMI. So does my zaptel
look correct? Maybe my cableing is off.
Thanks,
-Scott
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Friday, April 22, 2005 8:34 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
Of course there are exceptions to the rules. I see now on a couple
software releases where they do allow PRI with D4/AMI and PRI with
esf/b8zs. It's been a year or so since I messed with trunking on a 200,
I've mostly been installing and maintaining the SX2000's and 3300's.
Henry
- Original Message -
From: Dennis Walker [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, April 22, 2005 9:13 PM
Subject: RE: [Asterisk-Users] TE11OP - Mitel 200Sx??
I have done the same thing with an sx200 and a pri circuit
My sx200 can only do ami d4 and em channels
Here's parts of my config that takes the pri and converts it to em with
ANI DNIS
zaptel.conf
# t1 connected to the PRI circuit
span=1,1,0,exf,b8zs
# t1 connected to SX200
# the t1 card on my sx200 did d4 ami and I supplied ANI and DNIS through
the dial plan
span=2,0,0,d4,ami
bchan=1-23
dchan=24
em=25-47
-
zapata.conf
[channels]
echocancel=yes
echocancelwhenbridged=yes
echotraining=no
rxgain=0.0
txgain=0.0
useincomingcalleridonzaptransfer=yes
restrictcid=no
context=default
usecallingpres=yes
usercallerid=yes
hidecallerid=no
callerid=Company Name8005551212
signalling=pri_cpe
switchtype=dms100
group=1
channel = 1-23
group=2
signalling=em_w
emdigitwait=500
channel = 24-47
# I needed the emdigitwait=500 to wait long enough for the SX200 to dial
out it's digits
--
extensions.conf
# our PRI circiut gave us the last 4 digits of the dialed number and this
is how I passed
# *ANI*DNIS* to the SX200 for it to decode
# the first group were individual numbers that mapped to faxes and modems
exten = 1234,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = ,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = ,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
# this set mapped our did 5000 - 5199 to the SX200
exten = _5[0-1]XX,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
The reset of the dial plan took what ever I set up in the sx200 ARS to
dial
out and
sent out put Zap/G1
Hope this helps
--
From: Henry Devito[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, April 22, 2005 8:56 PM
To: Scott Wolfe; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
File: ATT00262.htmlFile: ATT00263.txt
I was wrong. I just looked in my Mitel IM's. What level software are
you
on in the SX200? Up until a certain level 200's could only do D4/AMI
T1's,
they could not do PRI's. If it is a newer switch within the past 3 years
or an older switch with later software than you can do PRI, but the
signaling and framing must be ESF/B8ZS.
Henry
- Original Message -
From: Scott Wolfe
To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial
Discussion
Sent: Friday, April 22, 2005 7:04 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
Thanks,
This is what I have now, but my Mitel PBX and Asterisk Box are unable
to communicate via the T1 connection. Asterisk loads ok but I get error
lights (blinking orange) on my TE110P and on my Mitel T1 card. Hu
-Scott
/etc/zaptel.conf
loadzone = us
defaultzone=us
span=1,0,0,d4,ami
bchan=1-23
dchan=24
/etc/asterisk/zapata.conf
[trunkgroups]
[channels]
context=default
switchtype=dms100
rxwink=300
usecallerid=no
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0 ;into the pstn twords the telco
txgain=0.0
callgroup=1
pickupgroup=1
immediate=yes
signalling=pri_cpe
group=1
context=default
emdigitwait=500
channel = 1-23 ; Set this to 1-15,17-31 for E1
- Original Message -
From: Michael D Schelin
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, April 22, 2005 4:48 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
Hello Henry
em=1-23 should be bchan=1-23
you have it set for analog
also
signaling=pri_cpe
Henry Devito wrote:
Don't you need one of these directives so the PRI knows which is
master and which is slave?
a.. pri_cpe: PRI signaling, CPE side
a.. pri_net: PRI signaling, Network side
Henry
- Original
Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
I have tried several, dlink doesn't seem to have the same issue and a more intelligent firewall is not having any problems. We are working with the Sipura 1001 and 2000 units on this issue. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Is your problem on the same model of Linksys? WRT54G? I haven't had a chance to try some other Linksys routers so I'm curious. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Scott Henderson Sent: Saturday, April 23, 2005 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Please make sure you post any solution you find to this issue to the list I have been frustrated by this as well. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri="sip:asterisk.mydomain.com", ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri="sip 66.x.x.166:5060", ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP registration behind Linksys WRT54G
I've got a 7960 behind a Linksys wireless box and its working just fine with nat=yes in the sip.conf. Has been for over a year. Not sure of the model though. Is your problem on the same model of Linksys? WRT54G? I haven't had a chance to try some other Linksys routers so I'm curious. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Henderson Sent: Saturday, April 23, 2005 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Please make sure you post any solution you find to this issue to the list I have been frustrated by this as well. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri=sip:asterisk.mydomain.com, ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri=sip 66.x.x.166:5060, ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. ___ Asterisk-Users mailing list Asterisk-Users@lists.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 registration behind Linksys WRT54G
The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --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] SIP registration behind Linksys WRT54G
I have a whole Asterisk server behind a wtr54gs. We have SPA-2000's registering from the Internet into it with no problems. Actually, we don't have it at the moment but did for several months. Not sure if this helps any or just adds to the confusion. - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 10:24 PM Subject: RE: [Asterisk-Users] SIP registration behind Linksys WRT54G I've got a 7960 behind a Linksys wireless box and its working just fine with nat=yes in the sip.conf. Has been for over a year. Not sure of the model though. Is your problem on the same model of Linksys? WRT54G? I haven't had a chance to try some other Linksys routers so I'm curious. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Henderson Sent: Saturday, April 23, 2005 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Please make sure you post any solution you find to this issue to the list I have been frustrated by this as well. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri=sip:asterisk.mydomain.com, ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri=sip 66.x.x.166:5060, ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] SIP registration behind Linksys WRT54G
Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --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 ___ Asterisk-Users mailing list Asterisk-Users@lists.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 registration behind Linksys WRT54G
I'm trying to register BT100s ... (doesn't work) X-Lite seems to work though Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo Sent: Saturday, April 23, 2005 8:48 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --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 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] SIP registration behind Linksys WRT54G
Have you tried to enable NAT translation on the Grandstream? On 4/23/05, Tomas Florian [EMAIL PROTECTED] wrote: I'm trying to register BT100s ... (doesn't work) X-Lite seems to work though Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo Sent: Saturday, April 23, 2005 8:48 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local
When using bristuff I do get an error too if I don't load zaptel first but not with the tdm driver. I know that in my modprobe.conf it is specified that ztcfg should be run after loading the module but why doesn't it? For some reason ztcfg is only 'accepted' when run from the cli Thanks! On Sat, 23 Apr 2005, Chris wrote: *sigh* I always get an error if I don't. Regards, Chris - Original Message - From: Eric Wieling aka ManxPower [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 7:15 PM Subject: Re: [Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local Chris wrote: You need this before wcfxs /sbin/modprobe zaptel *sigh* zaptel will automatically load when the card driver loads. modporbe will also run ztcfg after loading the card driver because (if you ran make install) /etc/modules.conf tells it to do so. -- 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
Re: [Asterisk-Users] ignorepat doesn't work
Jerry, when you say digitmap, you mean in my extensions.conf file? Thanks. Jaime From: Jerry [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Date: Sat, 23 Apr 2005 19:44:20 -0500 Try adding a comma to your digitmap where you wish the dialtone to come back on. Works on a Polycom. On Apr 23, 2005, at 7:12 PM, Eric Wieling aka ManxPower wrote: Grandstream does not support a dialplan. It is supposed to support Early Dial, but didn't work. I've been told that recent firmware fixes the early dial bug. I doubt that Early Dial is the solution. The solution is to buy a good IP Phone. Polycom and SIPura both support continue dialtone after digit. Cisco ATAs do not. I don't know if the Cisco IP phones do or not. Alexander Lopez wrote: ignorepat is for Zapata devices. Sip devices sned the number to the swith AFTER the SIP device feels it has dialed it. I am not a pro on the GS phones, (never played with them) but I would cheak the documentation on setting up a 'dialplan'. I hope this sets you in the right direction. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaime Blanco Sent: Saturday, April 23, 2005 4:38 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the ignorepat in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. -- 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
RE: [Asterisk-Users] SIP registration behind Linksys WRT54G
Yes that's the first thing I tried ... I'm able to make it work (using different routers than Linksys) in the following ways: - Set outgoing proxy and no STUN OR - No outgoing proxy and set STUN But once I put it behind Linksys everything registration does not work any more. Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Saturday, April 23, 2005 10:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Have you tried to enable NAT translation on the Grandstream? On 4/23/05, Tomas Florian [EMAIL PROTECTED] wrote: I'm trying to register BT100s ... (doesn't work) X-Lite seems to work though Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo Sent: Saturday, April 23, 2005 8:48 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best of the best of IP Phones
Ron Wellsted wrote: I have to agree, the Cisco 7960 is probably the best (I have yet to try a 7970/71). Cisco are a pain to deal with (they only want to deal with large value customers/distributors) and the phone do have some small quirks/bugs but they are the best in functionality and build quality. They are also the best speaker phone for small conferences. Have you tried Polycom? -- 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
