[Asterisk-Users] IVR Applications
Hello, Could someone please help refer me to a resource where I can find material on how to write IVR applications. I am using [EMAIL PROTECTED] ver. 2.8. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI to read MySQL
Hello everyone, I am not an expert in Asterisk programming yet. So, can someone help me put my first steps on how to use AGI to access MySQL tables and do queries. Any reference or help is appreciated. My target is to get Festival to read TTS from data stored in MySQL table based on the ID that the caller will input after his call is answered. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI not working
Hi everyone, I noticed that the waiting message indicator does not lit when I have a message in my voice mail. Any suggestion why this is happening? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI to read MySQL
Thanks a lot Fred. I will give it a try. I am using [EMAIL PROTECTED] V2.8 by the way. Regards Walid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Frederic JeanSent: Thursday, June 15, 2006 3:24 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] AGI to read MySQL Walid, Check the ASTCC agi script ; it just does exactly that: http://www.voip-info.org/wiki-ASTCC Cheers, Fred - Original Message - From: Walid Azab To: asterisk-users@lists.digium.com Sent: Thursday, June 15, 2006 11:14 Subject: [Asterisk-Users] AGI to read MySQL Hello everyone, I am not an expert in Asterisk programming yet. So, can someone help me put my first steps on how to use AGI to access MySQL tables and do queries. Any reference or help is appreciated. My target is to get Festival to read TTS from data stored in MySQL table based on the ID that the caller will input after his call is answered. Thanks ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: TTS from MySQL
Hi all, I need to simply use Asterisk to receive incoming calls in an IVR manner. It should authenticate users and read data from MySQL table that match their ID through Text-to-speech. I already have Asterisk 2.6 ([EMAIL PROTECTED]). I understand that I need to use Festival and AGI but do not know what to do exactly. Any help is appreciated. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TTS to read from Database
Hi everyone, Is there a way to get Asterisk read from MySQL using Festival Text to speech engine?!! Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: WEB SIP Dialer
Hi, I came across this nice looking web SIP dialer. However I cannot find how I can download it. Anyone know how...?? http://www.geocities.com/babarnazmi/ SIP (Session Initiation Protocol) based PC2Phone Dialer [more...http://www.angelfire.com/falcon/babarnazmi/SIPDialer/SIPDialer.htm]-PC2Phone (PC2PC) Dialer with latest SIP technology (SIP 2.0), NAT module(so it can operate easily through NAT and packet firewalls) with g723.1 codec and bandwidth control module. Customized stack for SIP protocol parsing. Microsoft and the Internet Engineering Task Force (IETF), have adopted the SIP technology as well as the Voice over IP community as its protocol of choice for signaling. [SIP Technology]. Compact and efficient (less than 650kb) Fully complies with SIP (RFC 3261), RTP/RTCP (RFC 1889), SDP (RFC 2327) PC-to-Phone, Phone-to-PC, PC-to-PC call models supported Local signalization (Dial tone, busy, ring back, etc.) for user comfort. Easy to install and configure NAT/Firewall support Low latency and adaptive jitter buffering Acoustic Echo Cancellation for speakerphone functions Voice Activity Detection for network bandwidth optimization Automatic Gain Control, self-adaptation of the microphone volume - no wizard needed Works with any full-duplex sound card Full integration of USB handset and headset devices, Builtin Actiontec support. Audio Tunning Wizard for setting sound/mic device and volume/Playback. Available on Windows 98, 98SE, Millennium, NT4, 2000, XP and XP+ Operating Systems ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Full T38 sip Faxing now Available
Hello, Could you provide me with more information on your solution. Thanks Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael D Schelin Sent: Thursday, July 28, 2005 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Full T38 sip Faxing now Available Hello everybody, for all of you that have searched for a real fax solution, look no further. We now have T38 faxing. Please contact me for more information. Thanks Michael D. Schelin ShellTel 626-814-2354 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Cisco 7940 - Disappearing Clock
I noticed this yesterday. I am using Cisco 7960 with SIP 7.4. It was my first time to see that in two days. I have no idea why this happened either. If you happen to know why, please drop me an e-mail. Thanks Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Thursday, July 28, 2005 3:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7940 - Disappearing Clock This question is not actually * related, but please don't flame me! Is anyone out there using the 7.4 or 7.5 SIP firmware on their Cisco 79xx phones? I have a weird problem where my clock disappears after a period of time, and the only thing that will get it back is a reboot. Has anyone experienced this? ___ Asterisk-Users mailing list Asterisk-Users@lists.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 WEB Phone (Wanna implement Click to Call)
Hi, I appreciate it if someone knows what is available for SIP web phones out there. I am interested in putting a soft phone on a website that registers with Asterisk using SIP. Then, when someone uses it, it directly calls into an asterisk call queue.. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] 7960 SIP Firmware Upgrade Strange Problem
Thanks for the heads up. I just followed [EMAIL PROTECTED] handbook. Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Wednesday, July 27, 2005 5:50 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem Hi On Tue, Jul 26, 2005 at 09:09:27PM +0200, Walid Azab wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Neil Cherry Sent: Tuesday, July 26, 2005 6:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem Walid Azab wrote: Thanks to all of you guys. I managed to fix it. It turned out to be that the ZIP file has to be extracted inside the TFTP root not outside then copied to the TFTP root. It is working now. Walid, you should be able to unzip it anywhere and copy it into the directory. It sounds like a permissions problem when you copied it. In the future just make sure that files copied into the tftp directory have at least read permission for everyone (user, group and other). Since it's working now you don't need to fool with it. Just information for the future. Yep you are right , I usually do a chmod 777. 755 would have been enough. 777 allows everyone who happen to get access to your network to change that firmware using simply tftp. Anyone feels like trojaning cisco phones? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.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] H323 Configuration file
Hi, This is what I have and is working just fine. I disabled Asterisk gatekeeper and registered directly to a Cisco CallManager 3.3.4 via h323 trunk. ; ; Configuration file of OpenH323 channel driver ; ;- ; General configuration options ; (ports, jitter, GK, ...) ;- [general] ; ; Address to bind to for incoming connections. ; Default is ALL. ; listenAddress=0.0.0.0 ; Port to listen to. ; Default value is 1720. ; listenPort=1720 ; ; Port to connect to. ; (Used only when we don't have a gatekeeper) ; Default value is 1720. ; connectPort=1720 ; ; Configure TCP port range to be used by H.323 ; tcpStart=1 tcpEnd=2 ; ; Configure UDP port range to be used by H.323 ; Note: The port range used by RTP are configured from ; rtp.conf ; udpStart=1 udpEnd=2 ; ; Enable fast start (yes,no). ; fastStart=no ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=no ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=no ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; inBandDTMF=no ; ; Enable silence suppression. ; silenceSuppression=no ; ; Set jitter buffer (in milliseconds, 20...1). ; jitterMin=20 jitterMax=100 ; ; Set IP Type-of-Service byte for RTP channels. ; Valid values for this option are: ; lowdelay, throughput, reliability, mincost, none ; ipTos=none ; ; Set the maximum number of inbound/outbound/simultaneous ; H.323 connections. ; outboundMax=10 inboundMax=10 simultaneousMax=10 ; ; Set the bandwidth limit for H.323 connections. ; The value is in Kbps. ; bandwidthLimit=1024 ; ; Set tracing options for the wrapper library and for the ; OpenH323 library. ; libTraceFile can be 'stdout' or a full path name to the tracefile. ; Only trace info for OpenH323 is logged in libTraceFile. ; wrapLibTraceLevel=3 libTraceLevel=3 libTraceFile=/root/h323.log ; ; Disable gatekeeper or specify a gatekeeper. ; Valid values for this option are: ; DISABLE, ; DISCOVER, ; gatekeeper's DNS name, ; gatekeeper's ip, ; GKID:gatekeeper's id ; ;gatekeeper=192.168.2.2 gatekeeper=DISABLE AllowGKRouted=yes ; ; Set the gatekeeper password ; ;gatekeeperPassword=secret ; ; Set the gatekeeper registration timeout ; gatekeeperTTL=600 ; ; Set the mode for sending user-input ; Valid values for this option are: ; Q931- Q.931 Keypad Information Element ; STRING - H.245 string ; TONE- H.245 tone ; RFC2833 - RFC2833 ; userInputMode=TONE ; ; AMA flags (default, omit, billing, documentation) ; amaFlags=default ; ; Account code ; accountCode=H323 ; ; Set the default context of H.323 calls. ; ;context=voip-h323 context=from-pstn ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; alias=asterisk alias=123 ; ; Aliases/prefixes routed in all-aliases context. ; context=all-aliases alias=ASTERISK alias=666 ; ; Aliases/prefixes routed in more-aliases context. ; context=more-aliases alias=665 ; ; Aliases/prefixes routed in all-prefixes context. ; context=all-prefixes gwprefix=00 gwprefix=01 ; ; Aliases/prefixes routed in more-stuff context. ; ;context=from-pstn ;alias=fax ;gwprefix=14002 ;- ; Specify and configure CODEC related ; options ;- [codecs] ; ; Define the codec list of the channel driver. ; Every codec option may have a frames option ; associated with it. ; Valid values for the codec option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3- G.723.1(6.3k) ; G72315K3- G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726- G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ; G72640K - G.726(40k) ; G728- G.728 ; G729- G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; codec=G711A frames=20 ;codec=G711U ;frames=20 ;codec=GSM0610 ;frames=4 ;codec=G7231 ;frames=2 ;codec=G729 ;frames=2 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Wednesday, July 27, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] H323 Configuration file Folks! I would appreciate if someone could send me a simple working h323 configuration file oh323.conf that is part of [EMAIL PROTECTED] installation. I have tried to use the oh323.conf content listed on WIKI but it is
[Asterisk-Users] Dial through IAX to FWD
Hi.. I am trying to do something but it is giving me some hard time here. I have an IAX2 trunk to FWD which is registered and working just fine. I have= 011|. as my dial pattern to allow that. But if I want to dial a toll free number I would have to dial 011*1800XXX What trunk dial rule should I use to enable anyone to call a toll free number by simply dialing 1800XX instead of having to dial 011*1800XXX? 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] Recording suddenly stopped
Hi..
I noticed all
recording activities suddenly stopped. It seems as if Asterisk is unable to
manipulate files. Here is a sample of a session in which I dialed the Voice Mail
system and tried to record my name:
Any
ideas?
Thanks
Executing
VoiceMail("SIP/100-69a9", "[EMAIL PROTECTED]") in
new stack -- Playing 'vm-theperson' (language
'en') -- Playing 'digits/1' (language
'en') -- Playing 'digits/0' (language
'en') -- Playing 'digits/0' (language
'en') -- Executing VoiceMailMain("SIP/100-69a9", "[EMAIL PROTECTED]") in new stack
-- Playing 'vm-password' (language 'en') -- Playing
'vm-youhave' (language 'en') -- Playing 'vm-no' (language
'en') -- Playing 'vm-messages' (language
'en') -- Playing 'vm-opts' (language
'en') -- Playing 'vm-starmain' (language
'en') -- Playing 'vm-opts' (language
'en') -- Playing 'vm-helpexit' (language
'en') -- Playing 'vm-options' (language
'en') -- Recording the message --
Playing 'vm-rec-name' (language 'en') -- Playing 'beep'
(language 'en') -- x=0, open writing:
voicemail/default/100/greet format: wav49, (nil)
-- Playing 'vm-review' (language 'en') -- Executing
Macro("SIP/100-69a9", "hangupcall") in new stack --
Executing ResetCDR("SIP/100-69a9", "w") in new stack --
Executing NoCDR("SIP/100-69a9", "") in new stack --
Executing Wait("SIP/100-69a9", "5") in new
stack
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[Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem
Hi, I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to 7.5 However in my first attempt to go from V.5.1 to 6.0 this is hat happens: - The phone reboots - The phone then reads the fileOS79XX.TXT from the TFP server. In the file I added the version "P0S3-06-0-00" - It starts upgrading firmware - Then I get the following message: (Upgrade Failed - Unauthorized) Any ideas? Please find below my conf files. SIP.CONF [300]username=300type=friendsecret=ciscorecord_out=On-Demandrecord_in=On-Demandqualify=noport=5060nat=never[EMAIL PROTECTED]host=dynamicdtmfmode=rfc2833context=from-internalcanreinvite=nocallerid="" 300 SIP000CCE351C07.cnf# SIP Configuration Generic File (start) # Line 1 Settingsline1_name: "300" ; Line 1 Extension\User IDline1_displayname: "300" ; Line 1 Display Nameline1_authname: "300" ; Line 1 Registration Authenticationline1_password: "cisco" ; Line 1 Registration Password # Line 2 Settingsline2_name: "" ; Line 2 Extension\User IDline2_displayname: "" ; Line 2 Display Nameline2_authname: "UNPROVISIONED" ; Line 2 Registration Authenticationline2_password: "UNPROVISIONED" ; Line 2 Registration Password # Line 3 Settingsline3_name: "" ; Line 3 Extension\User IDline3_displayname: "" ; Line 3 Display Nameline3_authname: "UNPROVISIONED" ; Line 3 Registration Authenticationline3_password: "UNPROVISIONED" ; Line 3 Registration Password # Line 4 Settingsline4_name: "" ; Line 4 Extension\User IDline4_displayname: "" ; Line 4 Display Nameline4_authname: "UNPROVISIONED" ; Line 4 Registration Authenticationline4_password: "UNPROVISIONED" ; Line 4 Registration Password # Line 5 Settingsline5_name: "" ; Line 5 Extension\User IDline5_displayname: "" ; Line 5 Display Nameline5_authname: "UNPROVISIONED" ; Line 5 Registration Authenticationline5_password: "UNPROVISIONED" ; Line 5 Registration Password # Line 6 Settingsline6_name: "" ; Line 6 Extension\User IDline6_displayname: "" ; Line 6 Display Nameline6_authname: "UNPROVISIONED" ; Line 6 Registration Authenticationline6_password: "UNPROVISIONED" ; Line 6 Registration Password # NAT/Firewall Traversalnat_address: ""voip_control_port: "5060"start_media_port: "16384"end_media_port: "32766" # Phone Label (Text desired to be displayed in upper right corner)phone_label: "WaZaB-SIP" ; Has no effect on SIP messaging # Time Zone phone will reside intime_zone: EST # Phone prompt/password for telnet/console sessionphone_prompt: "Cisco7960" ; Telnet/Console Promptphone_password: "abc" ; Telnet/Console Password # SIP Configuration Generic File (stop) SIPDefault.cnf # Image Versionimage_version: "P0S3-06-0-00" # Proxy Serverproxy1_address: "10.150.200.165"# Proxy Server Port (default - 5060)proxy1_port:"5060" # Emergency Proxy infoproxy_emergency: "10.150.200.165"proxy_emergency_port: "5060" # Backup Proxy infoproxy_backup: "10.150.200.165"proxy_backup_port: "5060"# Outbound Proxy infooutbound_proxy: ""outbound_proxy_port: "5060"# NAT/Firewall Traversalnat_enable: "0"nat_address: ""voip_control_port: "5061"start_media_port: "16384"end_media_port: "32766"nat_received_processing: "0" # Proxy Registration (0-disable (default), 1-enable)proxy_register: "1"# Phone Registration Expiration [1-3932100 sec] (Default - 3600)timer_register_expires: "3600"# Codec for media stream (g711ulaw (default), g711alaw, g729)preferred_codec: "none"# TOS bits in media stream [0-5] (Default - 5)tos_media: "5" # Enable VAD (0-disable (default), 1-enable)enable_vad: "0"# Allow for the bridge on a 3way call to join remaining parties upon hangupcnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default)# Allow Transfer to be completed while target phone is still ringingsemi_attended_transfer: "0" ; 0-Disabled, 1-Enabled (default)# Telnet Level (enable or disable the ability to telnet into this phone telnet_level: "2" ; 0-Disabled (default), 1-Enabled, 2-Privileged # Inband DTMF Settings (0-disable, 1-enable (default))dtmf_inband: "1"# Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt )dtmf_outofband: "avt"# DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)dtmf_db_level: "3"# SIP Timerstimer_t1: "500" ; Default 500 msectimer_t2: "4000" ; Default 4 secsip_retx: "10" ; Default 11sip_invite_retx: "6" ; Default 7timer_invite_expires: "180" ; Default 180 sec# Setting for Message speeddial to UOne boxmessages_uri: "*97" # TFTP Phone Specific Configuration File Directorytftp_cfg_dir: "./"# Time Serversntp_mode: "unicast"sntp_server: "10.150.200.165"time_zone: "EST"dst_offset: "1"dst_start_month: "April"dst_start_day: ""dst_start_day_of_week: "Sun"dst_start_week_of_month: "1"dst_start_time: "02"dst_stop_month: "Oct"dst_stop_day: ""dst_stop_day_of_week: "Sunday"dst_stop_week_of_month: "8"dst_stop_time: "2"dst_auto_adjust: "1"# Do Not Disturb Control (0-off, 1-on, 2-off with
RE: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem
Title: Message Thanks to all of you guys. I managed to fix it. It turned out to be that the ZIP file has to be extracted inside the TFTP root not outside then copied to the TFTP root. It is working now. Thanks Walid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Watkins, BradleySent: Tuesday, July 26, 2005 4:12 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem I believe you have to upgrade to 5.3 in order to go from unsigned to signed executables. Once you're at 5.3, you can go directly to 7.5. I did this recently with a couple of 7960s I had in the lab and it worked perfectly. Regards, - Brad -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walid AzabSent: Tuesday, July 26, 2005 10:29 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem Hi, I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to 7.5 However in my first attempt to go from V.5.1 to 6.0 this is hat happens: - The phone reboots - The phone then reads the fileOS79XX.TXT from the TFP server. In the file I added the version "P0S3-06-0-00" - It starts upgrading firmware - Then I get the following message: (Upgrade Failed - Unauthorized) Any ideas? Please find below my conf files. SIP.CONF [300]username=300type=friendsecret=ciscorecord_out=On-Demandrecord_in=On-Demandqualify=noport=5060nat=never[EMAIL PROTECTED]host=dynamicdtmfmode=rfc2833context=from-internalcanreinvite=nocallerid="" 300 SIP000CCE351C07.cnf# SIP Configuration Generic File (start) # Line 1 Settingsline1_name: "300" ; Line 1 Extension\User IDline1_displayname: "300" ; Line 1 Display Nameline1_authname: "300" ; Line 1 Registration Authenticationline1_password: "cisco" ; Line 1 Registration Password # Line 2 Settingsline2_name: "" ; Line 2 Extension\User IDline2_displayname: "" ; Line 2 Display Nameline2_authname: "UNPROVISIONED" ; Line 2 Registration Authenticationline2_password: "UNPROVISIONED" ; Line 2 Registration Password # Line 3 Settingsline3_name: "" ; Line 3 Extension\User IDline3_displayname: "" ; Line 3 Display Nameline3_authname: "UNPROVISIONED" ; Line 3 Registration Authenticationline3_password: "UNPROVISIONED" ; Line 3 Registration Password # Line 4 Settingsline4_name: "" ; Line 4 Extension\User IDline4_displayname: "" ; Line 4 Display Nameline4_authname: "UNPROVISIONED" ; Line 4 Registration Authenticationline4_password: "UNPROVISIONED" ; Line 4 Registration Password # Line 5 Settingsline5_name: "" ; Line 5 Extension\User IDline5_displayname: "" ; Line 5 Display Nameline5_authname: "UNPROVISIONED" ; Line 5 Registration Authenticationline5_password: "UNPROVISIONED" ; Line 5 Registration Password # Line 6 Settingsline6_name: "" ; Line 6 Extension\User IDline6_displayname: "" ; Line 6 Display Nameline6_authname: "UNPROVISIONED" ; Line 6 Registration Authenticationline6_password: "UNPROVISIONED" ; Line 6 Registration Password # NAT/Firewall Traversalnat_address: ""voip_control_port: "5060"start_media_port: "16384"end_media_port: "32766" # Phone Label (Text desired to be displayed in upper right corner)phone_label: "WaZaB-SIP" ; Has no effect on SIP messaging # Time Zone phone will reside intime_zone: EST # Phone prompt/password for telnet/console sessionphone_prompt: "Cisco7960" ; Telnet/Console Promptphone_password: "abc" ; Telnet/Console Password # SIP Configuration Generic File (stop) SIPDefault.cnf # Image Versionimage_version: "P0S3-06-0-00" # Proxy Serverproxy1_address: "10.150.200.165"# Proxy Server Port (default - 5060)proxy1_port:"5060" # Emergency Proxy infoproxy_emergency: "10.150.200.165"proxy_emergency_port: "5060" # Backup Proxy infoproxy_backup: "10.150.200.165"proxy_backup_port: "5060"# Outbound Proxy infooutbound_proxy: ""outbound_proxy_port: "5060"# NAT/Firewall Traversalnat_enable: "0"nat_address: ""voip_control_port: "5061"start_media_port: "16384"end_media_port: "32766"nat_received_processing: "0" # Proxy Registration (0-disable (default), 1-enable)proxy_register: "1"# Phone Registration Expiration [1-3932100 sec] (Default - 3600)timer_register_expires: "3600"# Codec for media stream (g711ulaw (default), g711alaw, g729)preferred_codec: "none"# TOS bits in media stream [0-5] (Default - 5)tos_media: "5" # Enable VAD (0-disable (default), 1-enable)enable_vad: "0"# Allow for the bridge on a 3way call to join remaining parties upon hangupcnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default)# Allow Transfer to be
[Asterisk-Users] 7960 from SIP to SKINNY
Hello, Anyone tried reverting to SKINNY from SIP. I have a problem I cannot fix and need to get back to SCCP to be able to use the phone. ThanksWalid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] 7960 SIP Firmware Upgrade Strange Problem
Title: Message I wentfrom 5.3 to 6.3 then from 6.3 t 7.5 directly. However, I have the warning message (Protocol Application Invalid) Please any help. Walid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Watkins, BradleySent: Tuesday, July 26, 2005 4:12 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem I believe you have to upgrade to 5.3 in order to go from unsigned to signed executables. Once you're at 5.3, you can go directly to 7.5. I did this recently with a couple of 7960s I had in the lab and it worked perfectly. Regards, - Brad -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walid AzabSent: Tuesday, July 26, 2005 10:29 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem Hi, I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to 7.5 However in my first attempt to go from V.5.1 to 6.0 this is hat happens: - The phone reboots - The phone then reads the fileOS79XX.TXT from the TFP server. In the file I added the version "P0S3-06-0-00" - It starts upgrading firmware - Then I get the following message: (Upgrade Failed - Unauthorized) Any ideas? Please find below my conf files. SIP.CONF [300]username=300type=friendsecret=ciscorecord_out=On-Demandrecord_in=On-Demandqualify=noport=5060nat=never[EMAIL PROTECTED]host=dynamicdtmfmode=rfc2833context=from-internalcanreinvite=nocallerid="" 300 SIP000CCE351C07.cnf# SIP Configuration Generic File (start) # Line 1 Settingsline1_name: "300" ; Line 1 Extension\User IDline1_displayname: "300" ; Line 1 Display Nameline1_authname: "300" ; Line 1 Registration Authenticationline1_password: "cisco" ; Line 1 Registration Password # Line 2 Settingsline2_name: "" ; Line 2 Extension\User IDline2_displayname: "" ; Line 2 Display Nameline2_authname: "UNPROVISIONED" ; Line 2 Registration Authenticationline2_password: "UNPROVISIONED" ; Line 2 Registration Password # Line 3 Settingsline3_name: "" ; Line 3 Extension\User IDline3_displayname: "" ; Line 3 Display Nameline3_authname: "UNPROVISIONED" ; Line 3 Registration Authenticationline3_password: "UNPROVISIONED" ; Line 3 Registration Password # Line 4 Settingsline4_name: "" ; Line 4 Extension\User IDline4_displayname: "" ; Line 4 Display Nameline4_authname: "UNPROVISIONED" ; Line 4 Registration Authenticationline4_password: "UNPROVISIONED" ; Line 4 Registration Password # Line 5 Settingsline5_name: "" ; Line 5 Extension\User IDline5_displayname: "" ; Line 5 Display Nameline5_authname: "UNPROVISIONED" ; Line 5 Registration Authenticationline5_password: "UNPROVISIONED" ; Line 5 Registration Password # Line 6 Settingsline6_name: "" ; Line 6 Extension\User IDline6_displayname: "" ; Line 6 Display Nameline6_authname: "UNPROVISIONED" ; Line 6 Registration Authenticationline6_password: "UNPROVISIONED" ; Line 6 Registration Password # NAT/Firewall Traversalnat_address: ""voip_control_port: "5060"start_media_port: "16384"end_media_port: "32766" # Phone Label (Text desired to be displayed in upper right corner)phone_label: "WaZaB-SIP" ; Has no effect on SIP messaging # Time Zone phone will reside intime_zone: EST # Phone prompt/password for telnet/console sessionphone_prompt: "Cisco7960" ; Telnet/Console Promptphone_password: "abc" ; Telnet/Console Password # SIP Configuration Generic File (stop) SIPDefault.cnf # Image Versionimage_version: "P0S3-06-0-00" # Proxy Serverproxy1_address: "10.150.200.165"# Proxy Server Port (default - 5060)proxy1_port:"5060" # Emergency Proxy infoproxy_emergency: "10.150.200.165"proxy_emergency_port: "5060" # Backup Proxy infoproxy_backup: "10.150.200.165"proxy_backup_port: "5060"# Outbound Proxy infooutbound_proxy: ""outbound_proxy_port: "5060"# NAT/Firewall Traversalnat_enable: "0"nat_address: ""voip_control_port: "5061"start_media_port: "16384"end_media_port: "32766"nat_received_processing: "0" # Proxy Registration (0-disable (default), 1-enable)proxy_register: "1"# Phone Registration Expiration [1-3932100 sec] (Default - 3600)timer_register_expires: "3600"# Codec for media stream (g711ulaw (default), g711alaw, g729)preferred_codec: "none"# TOS bits in media stream [0-5] (Default - 5)tos_media: "5" # Enable VAD (0-disable (default), 1-enable)enable_vad: "0"# Allow for the bridge on a 3way call to join remaining parties upon hangupcnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default)# Allow Transfer to be completed while target phone is still
RE: [Asterisk-Users] 7960 from SIP to SKINNY
I just succeeded in doing so. My problem was also with obtaining the Skinny image. But I managed to get one off the existing Cisco Call Manager that we have. Kind Regards, Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Tuesday, July 26, 2005 7:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 7960 from SIP to SKINNY On Tue, 2005-07-26 at 18:31 +0200, Walid Azab wrote: Hello, Anyone tried reverting to SKINNY from SIP. I have a problem I cannot fix and need to get back to SCCP to be able to use the phone. It works fine. You can edit your SIPxxx file and add a line: image_version:P00307010200 This will change it to sccp. You will need to then use the SEPxxx to change it back. -- respectfully, Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.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] 7960 SIP Firmware Upgrade Strange Problem
Yep you are right , I usually do a chmod 777. Thanks anyway :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Neil Cherry Sent: Tuesday, July 26, 2005 6:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem Walid Azab wrote: Thanks to all of you guys. I managed to fix it. It turned out to be that the ZIP file has to be extracted inside the TFTP root not outside then copied to the TFTP root. It is working now. Walid, you should be able to unzip it anywhere and copy it into the directory. It sounds like a permissions problem when you copied it. In the future just make sure that files copied into the tftp directory have at least read permission for everyone (user, group and other). Since it's working now you don't need to fool with it. Just information for the future. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ Asterisk-Users mailing list Asterisk-Users@lists.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 and Cisco CallManager Integration
Hi again, Well, thanks for the details steps. But before I received your mail I had already installed [EMAIL PROTECTED] v.1.3 and updated it with OH323 add-on. It is a zip file which when you install you get all the libraries installed and compiled for you. Now, one last step for me which I need your help all with. What is needed to get the CCM and Asterisk to exchange calls over H323? I mean which config files needs to be updated. I now have oh323.conf shown and ready. Thanks Walid Subject: Re: [Asterisk-Users] Asterisk and Cisco CallManager Integration From: Vamsi Pottangi [EMAIL PROTECTED] Date: Mon, 27 Jun 2005 11:16:49 +0530 Reply-to: Vamsi Pottangi [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] The below worked for me to integrate with CCM. pwlib-v1_6_6 openh323-v1_13_5 asterisk-oh323-0.7.1 The only change I made was -- Remove the line 433 (:protected) in /usr/src/openh323/include/gkserver.h else you would get the below error during compilation /usr/src/openh323/include/gkserver.h:434: error: `virtual H323Transaction::Response H323GatekeeperRRQ::OnHandlePDU()' is protected -- Steps to follow: --- To enable H323 for inter-op with Cisco Call Manager (H.323) cp pwlib-v1_6_6-src.tar.gz openh323-v1_13_5-src.tar.gz asterisk-oh323-0.7.1.tar.gz /usr/src/ cd /usr/src tar zxf pwlib-v1_6_6-src.tar.gz tar zxf openh323-v1_13_5-src.tar.gz tar zxf asterisk-oh323-0.7.1.tar.gz - Set Environment variables PWLIBDIR=/usr/src/pwlib OPENH323DIR=/usr/src/openh323 LD_LIBRARY_PATH=/usr/src/pwlib/lib:/usr/src/openh323/lib cd /usr/src/pwlib ./configure make opt cd /usr/src/openh323 ./configure -- Remove the line 433 (:protected) in /usr/src/openh323/include/gkserver.h else you would get the below error during compilation /usr/src/openh323/include/gkserver.h:434: error: `virtual H323Transaction::Response H323GatekeeperRRQ::OnHandlePDU()' is protected -- make opt cd /usr/src/asterisk-oh323-0.7.1 Edit makefile and set the paths/options according to your system. Type "make" to build the oh323wrap library and the ASTERISK OH323 channel driver. - If compiling fails, then change the makefile to reflect the below CPPFLAGS=$(OPENH323FLAGS) -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -Wall -fPIC -I/usr/src/pwlib/include -DPTRACING -I/usr/src/openh323/include -DHAS_OSS -Wall -x c++ -Os --- Type "make install" to install the binaries. This will also install a sample configuration file, if there isn't one. Next, add to your LD_LIBRARY_PATH the path where the oh323wrap library was installed (or edit your /etc/ld.so.conf file, add the library path, and run "ldconfig"). Thanks, ~Vamsi On 6/26/05, Walid Azab [EMAIL PROTECTED] wrote: I have previously tried the Asterisk/OH323/PWLIB/GNUGK combination and had problems compiling OH323. I will try again from a clean installation. On the other hand, can you send me any useful links or guides that you already used. This can make our trial and error efforts much less. Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Greg Oliver Sent: Sunday, June 26, 2005 2:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Cisco CallManager Integration We have successfully connect * .9x 1.0.x with CCM 3.3.x and up using both gatekeeper and no gatekeeper.. Using SIP usually with CCM 4.0 and up.. With CCM 3.3.x, there is a limitation where the gateway H323 in your case cannot use IP addresses, so the Asterisk box has to have correct DNS entries to resolbve your asterisk ox.. Then just use regular route patterns and direct it to asterisk.. That works well. You may also want to make sure your compatibility matrix between Asterisk/OH323/PWLIB/GNUGK is right - incompatibilities cause more issues than I care to talk about. The GNUGk web site has the best matrix to follow.. Thanks, GReg On Sat, 2005-06-25 at 10:39 -0500, [EMAIL PROTECTED] wrote: Use a gatekeeper and have both boxes register with the gatekeeper. That way you can specify what numbers go where. From everything I have tested, * will NOT register with CCM. When I added in a gatekeeper and had both sides register with it, everything works. Walid Azab wrote: Hello, I have Cisco CallManager 3.3.4 and [EMAIL PROTECT
[Asterisk-Users] Asterisk on Linksys WRT54G
Hi
all,
Any one tried
installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to
SIP calls. The phones ring and calls are established but we cannot hear any
voice at all. I tried allow=all in the general section but did not work. So I
forced ulaw. Can any one please check it out and let me know what is
wrong?
Here are the conf
files:
Asterisk
Version: Asterisk CVS-HEAD-01/17/05-00:35:58 built by [EMAIL PROTECTED] on a i686 running
Linux
==SIP.CONF
[general]
port =
5060 ; Port to bind
to (SIP is 5060)bindaddr = 0.0.0.0 ; Address to bind to
(all addresses on
machine)disallow=all
; Allow all codecsallow=ulawcontext = bogon-calls ; Send SIP callers
that we don't know about here
[2000]
type=friend
; This device takes and makes
callsusername=2000 ;
Username on
devicesecret=1234
; Password for
devicehost=dynamic ;
This host is not on the same IP addr every
timecontext=from-sip ; Inbound calls from this
host go
heremailbox=100
; Activate the message waiting light if
this
; voicemailbox has messages in it
[2001]
; Duplicate of 2000, except with different auth data
type=friendusername=2001secret=1234host=dynamiccontext=from-sipmailbox=101
==Extensions.conf
[general]
static=yeswriteprotect=yes
[bogon-calls]
exten =
_.,1,Congestion
[from-sip]
exten =
2000,1,Dial(SIP/2000,20)
exten =
2000,2,Voicemail(u2000)
exten =
2000,102,Voicemail(b2000)exten = 2000,103,Hangup
exten =
2001,1,Dial(SIP/2001,20)exten = 2001,2,Voicemail(u2001)exten =
2001,102,Voicemail(b2001)exten = 2001,103,Hangup
exten =
2999,1,VoicemailMain(${CALLERIDNUM})
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RE: [Asterisk-Users] Asterisk and Cisco CallManager Integration
I have previously tried the Asterisk/OH323/PWLIB/GNUGK combination and had problems compiling OH323. I will try again from a clean installation. On the other hand, can you send me any useful links or guides that you already used. This can make our trial and error efforts much less. Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Oliver Sent: Sunday, June 26, 2005 2:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Cisco CallManager Integration We have successfully connect * .9x 1.0.x with CCM 3.3.x and up using both gatekeeper and no gatekeeper.. Using SIP usually with CCM 4.0 and up.. With CCM 3.3.x, there is a limitation where the gateway H323 in your case cannot use IP addresses, so the Asterisk box has to have correct DNS entries to resolbve your asterisk ox.. Then just use regular route patterns and direct it to asterisk.. That works well. You may also want to make sure your compatibility matrix between Asterisk/OH323/PWLIB/GNUGK is right - incompatibilities cause more issues than I care to talk about. The GNUGk web site has the best matrix to follow.. Thanks, GReg On Sat, 2005-06-25 at 10:39 -0500, [EMAIL PROTECTED] wrote: Use a gatekeeper and have both boxes register with the gatekeeper. That way you can specify what numbers go where. From everything I have tested, * will NOT register with CCM. When I added in a gatekeeper and had both sides register with it, everything works. Walid Azab wrote: Hello, I have Cisco CallManager 3.3.4 and [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] latest version. I have earlier tried getting Asterisk to register with CCM via H323 and failed. Back then, I learned that this is a known bug in Asterisk. Also people who tried doing that had also succeeded in getting calls to go through only one direction like from CCM to Asterisk. I am not that expert so excuse my ignorance with this subject. So please if anyone has any useful information or is sure that this can now work please send me whatever you have on that. I simply want Asterisk users to get their dial tones through CCM. Thanks and I appreciate your assistance. Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] FW: ZAP to SIP Dial Plan
Hi, I would like to setup Asterisk to route incoming calls to ZAP on my TDM400P to SIP phones. What is the best dial plan to use. We are currently able to route outgoing calls to PSTN from SIP to ZAP. Thanks in advance. Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Asterisk and Cisco CallManager Integration
Hello, I have Cisco CallManager 3.3.4 and [EMAIL PROTECTED] latest version. I have earlier tried getting Asterisk to register with CCM via H323 and failed. Back then, I learned that this is a known bug in Asterisk. Also people who tried doing that had also succeeded in getting calls to go through only one direction like from CCM to Asterisk. I am not that expert so excuse my ignorance with this subject. So please if anyone has any useful information or is sure that this can now work please send me whatever you have on that. I simply want Asterisk users to get their dial tones through CCM. Thanks and I appreciate your assistance. Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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 to ZAP Dialplan
Hi, I would like to setup Asterisk to route incoming calls to ZAP on my TDM400P to SIP phones. What is the best dial plan to use. We are currently able to route outgoing calls to PSTN from SIP to ZAP. Thanks in advance. Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Install Asterisk on CCM MCS-7835 Server
Hi All, I am replacing Cisco Call Manager with Asterisk. As you know CCM is on a MCS 7835 Server which comes with a custom version of Windows. Does any one know how to install Linux on that H/W. My guess is that someone must have tried the same thing before. I know how to install Linux however I cannot get passed the H/W limitation. Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Install Asterisk on CCM MCS-7835 Server
Hi All, I am replacing Cisco Call Manager with Asterisk. As you know CCM is on a MCS 7835 Server which comes with a custom version of Windows. Does any one know how to install Linux on that H/W. My guess is that someone must have tried the same thing before. I know how to install Linux however I cannot get passed the H/W limitation. Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] FW: SIP Phone Choices
Hi, What are the SIP phone models that proved to be working well with Asterisk? I appreciate your recommendations. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Asterisk and SER
Hi Everyone, Just a curious question. Has anyone heard of any service provider who is using Asterisk and SER to provide their VOIP services? Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Analogue Line to Asterisk (Which Digium Model???)
Guys, I need to use Asterisk to call out PSTN numbers via an analogue line. I understand Digium manufactures these kinds of cards, but can someone tell me which model number it is. I really only need a card with one or 2 analogue ports max. Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Asterisk Versioning
Hi, Just want to understand the difference between Asterisk Versions and please correct me if I am wrong, I understand they are: Stable CVS CVS Head I am a newbie and about to install Asterisk on SUSE Server. Can someone please advise what is the best version type and number should I use. My environment is not so big. I only wish to eventually get my asterisk to talk to Cisco CCM 3.3.4. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Bellster - cool :-)
Bellster.net say that you can: -put a 'q' in front of the number that you are calling. e.g. 'q12125551212' -- Any one knows how you can do that using the phone dialpad? Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan Goodwin Sent: Sunday, January 23, 2005 2:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bellster - cool :-) I love belster, I added a route for the 518 area code, (that covers most of upstate NY), only thing I wish I could do is get rid of the message that says how many credits I have left. I would rather it just report congested is the call can't go though (doto lack of credits), that way I could make Bellster my default route, then use another if it doesn't work as a backup. I made a few test calls to different places using Bellster, surprizingly the quility was very good. Steven P. Donegan wrote: OK, I have done all the stuff at my end and at Bellsters end to add 21 new area codes (all of california) to the Bellster dial plan. Pretty cool deal! I hope others go for this quickly - as it could be a really nice co-op. I do suggest to Jeff - do some sort of calling trunk -vs- routed trunk match to make sure that someone can't run their credits sky-high by making calls through themselves. I did all my test calls through my own trunks and voila I have credits available. Jeff - you rock :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.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] H323 Softphone for iPAQ
Since I want the PDAs to talk to Cisco CallManager, I think I should better look for Skinny pocket pc clients. Isn't that correct! Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Monday, January 17, 2005 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 Softphone for iPAQ Also the following has worked great for me: http://www.wifive.net/introduction.asp Michael Radovan Mihalik wrote: http://www.sjlabs.com/sjp.html SJphoneR is a VOIP softphone that allows you to speak with any PC, PDA, stand-alone IP-phone and with any legacy wired or mobile phone (using your VOIP gateway or purchasing service from Internet Telephony Service Provider). It supports both SIP and H.323 standards and is fully interoperable with most major IP-telephony vendors and ITSP. I'm just about to try it my self ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walid Azab Sent: Sunday, January 16, 2005 8:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] H323 Softphone for iPAQ Hi list, I was just wondering, is there any H.323 soft-phone that can be installed on a pocket PC (iPAQ). Walid -- -- ___ Asterisk-Users mailing list Asterisk-Users@lists.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] H323 Softphone for iPAQ
Hi list, I was just wondering, is there any H.323 soft-phone that can be installed on a pocket PC (iPAQ). Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Asterisk to CCM3.3.4 via H32
Hi.. I need to send calls coming from SIP phones behind asterisk to Cisco Call Manager 3.3.4. We have created an H323 trunk on the call manager.Provided that Asterisk-oh323 is installed, how should h323.conf be configured for that? Also when this is done can I setup CCM to alert phones behind asterisk when it receives a call from PSTN?, provided that PSTN to CCM is configured and working. 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] oh323 compile error
I am trying to compile oh323 and having the following error. Can anyone help please?! This is my third post. These are the versions I am using: Compilation Error: -- g++ -o obj_linux_x86_r/simph323 -s -L/root/pwlib/lib -L/root/openh323/lib ./obj_linux_x86_r/main.o -lh323_linux_x86_r -lpt_linux_x86_r -lpthread -lssl -lcrypto -lexpat -lresolv -ldl/root/openh323/lib/libh323_linux_x86_r.so: undefined reference to `std::basic_iostreamchar, std::char_traitschar ::~basic_iostream()'/root/pwlib/lib/libpt_linux_x86_r.so: undefined reference to `std::basic_iostreamchar, std::char_traitschar ::basic_iostream(std::basic_streambufchar, std::char_traitschar *)'/root/pwlib/lib/libpt_linux_x86_r.so: undefined reference to `std::basic_iostreamchar, std::char_traitschar ::~basic_iostream()'/root/openh323/lib/libh323_linux_x86_r.so: undefined reference to `std::basic_iostreamchar, std::char_traitschar ::~basic_iostream()'collect2: ld returned 1 exit statusmake[1]: *** [obj_linux_x86_r/simph323] Error 1make[1]: Leaving directory `/root/openh323/samples/simple'make: *** [opt] Error 2 Files Versions used: -- 1-openh323-Janus_patch4-src-tar.gz ==from http://sourceforge.net/projects/openh323 (v1.13.5) 2- pwlib-Janus_patch4-src-tar.gz == from http://sourceforge.net/projects/openh323(v1.6.6) 3. asterisk-oh323-0.6.5.tar.gz== from http://www.inaccessnetworks.com/projects/asterisk-oh323/download (v1.6.5) 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] Add h323 support to Asterisk
I have asterisk CVS-v1-0-12 Can someone please advise what is the best solution and versions for adding h323 support to asterisk. I am confused between oh323/pwlib/asteris-oh323 versions. Asterisk oh323 0.7 README say I need to getPWlib (v1.6.6) and OpenH323 (v1.13.5) but I cannot find them. Please help. Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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 on a notebook?
You might need to go for [EMAIL PROTECTED] Avery simple and easy to install version of Asterisk. Just burn the ISO image to a CD and boot with it and it will automatically install everything for you. However, it will wipe out all your HD and install CentOS then Asterisk. For SIP, you can start right away by using X-lite (SoftPhone) or any SIP IP phone. Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Thursday, January 13, 2005 6:09 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk on a notebook? I'd dearly love to be able to give an Asterisk demo by just toting my notebook, a PC/PCMCIA card, and a couple SIP phones. Is there any way to do this? Or should I look for a small-profile box with PCI slots, instead? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Cisco 79XX phones not talking to asterisk
If there is a little X next to the line icon then the phone is not registered. Try showing registered phones from asterisk console using SIP Show peers It will tell you which phones are registered with which IPs. Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Thursday, January 13, 2005 6:54 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 79XX phones not talking to asterisk Hi all, I have setup my Cisco 79XX phone. Did the tftp, put the config files in the right location with the right names. Booted my phone, it does the tftp things, the screen shows my extensions everything seems fine. However, when I come offhook and try to dial 11 which is just a playback of demo-congrats in the dialplan the phone says Calling Out (INV) below is my sip.conf file - I presume it is not correcly registering with asterisk. The phone boots DHCP gets an address, loads the SIP software and sets there for me to dial. However, I get the INV when I dial. Any ideas on why the phone is displaying invalid and what to do about it??? Thanks, jerry sip.conf [201] type=friend dtmfmode=rfc2833 username=201 secret=201 disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid=Media Assistant 201 [202] type=friend dtmfmode=rfc2833 username=202 secret=202 disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid=Media Assistant 201 [203] type=friend dtmfmode=rfc2833 username=203 secret=203 disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid=Media Assistant 201 [204] type=friend dtmfmode=rfc2833 username=204 secret=204 disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid=Media Assistant 201 [205] type=friend dtmfmode=rfc2833 username=205 secret=205 disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid=Media Assistant 201 [206] type=friend dtmfmode=rfc2833 username=206 secret=206 disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid=Media Assistant 201 extension.conf [smvoice-sip] exten = 11,1,Playback(demo-congrats) exten = 11,2,Hangup ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Want to install Oh323 and LOST
I need to install Oh323 in order to get Asterisk connect to Cisco CCM 3.3.4. I cannot find the dependencies needed for that. Can any one send me link: The readme file says they areat the following links but the exact version isn't. Please help. Walid o PWlib (Portable Text and GUI C/C++ Class Library) download from http://sourceforge.net/projects/openh323 (v1.6.6) (required) o OpenH323 (Class Library implementing the H.323 protocol) download from http://sourceforge.net/projects/openh323 (v1.13.5) (required) ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Build PWLIB
I am trying to build PWLIB to get OH323 up and running. I am not an expert in linux but can someone help telling me how I can do the following: How can I add a directory to LD_LIBRARY_PATH?! Thanks in advance --For unix.-- 1. If you have not put pwlib it into your home directory (~/pwlib) then you will have to defined the environment variable PWLIBDIR to point to the correct directory. Also make sure you have added the $PWLIBDIR/lib directory to your LD_LIBRARY_PATH environment variable if you intend to use shared libraries (the default). - ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Oh323 compilation errors
Hi, well, I really need your help here. I have tried compiling oh323 many times and I always get the following error when trying to "make opt" open h323. Any ideas?! Compilation Error: -- g++ -o obj_linux_x86_r/simph323 -s -L/root/pwlib/lib -L/root/openh323/lib ./obj_linux_x86_r/main.o -lh323_linux_x86_r -lpt_linux_x86_r -lpthread -lssl -lcrypto -lexpat -lresolv -ldl/root/openh323/lib/libh323_linux_x86_r.so: undefined reference to `std::basic_iostreamchar, std::char_traitschar ::~basic_iostream()'/root/pwlib/lib/libpt_linux_x86_r.so: undefined reference to `std::basic_iostreamchar, std::char_traitschar ::basic_iostream(std::basic_streambufchar, std::char_traitschar *)'/root/pwlib/lib/libpt_linux_x86_r.so: undefined reference to `std::basic_iostreamchar, std::char_traitschar ::~basic_iostream()'/root/openh323/lib/libh323_linux_x86_r.so: undefined reference to `std::basic_iostreamchar, std::char_traitschar ::~basic_iostream()'collect2: ld returned 1 exit statusmake[1]: *** [obj_linux_x86_r/simph323] Error 1make[1]: Leaving directory `/root/openh323/samples/simple'make: *** [opt] Error 2 Files Versions used: -- 1-openh323-Janus_patch4-src-tar.gz 2- pwlib-Janus_patch4-src-tar.gz 3. asterisk-oh323-0.6.5.tar.gz Installation steps: -- Install Pwlib#cd pwlib#./configure make clean make opt make install ldconfigPatch and Install OpenH323#cd openh323#patch -p1 ../asterisk-oh323-0.6.5/openh323_1.13.5-make.patch#./configure make clean make opt make install ldconfigAsterisk#cd asterisk-1.0.3#make make install make samplesAsterisk-oh323#cd asterisk-oh323-0.6.5Edit the Makefile#make make install ldconfig !! :| ___ Asterisk-Users mailing list Asterisk-Users@lists.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] What's the easiest way to get * to call PSTN?
Hi, I just want to know what is the easiest way to have Asterisk route calls to PSTN. Hope any one can help me. PS: Any solution using a Cisco device is preferable. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] H323 on Asterisk@Home
Guys, I am about to install H323 [EMAIL PROTECTED] (Asterisk CVS-v1-0-12/22/04-05:48:41). I noticed that the default h323h.conf file is not set up. I also noticed that many of you here say that it is better to use Oh323. What is the best scenario here for me? Should I go with the already existing h323 located on (channels/h323)? or go for oh323? BTW, how can I get PWLib! Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [Asterisk-Users] asterisk - oh323 driver
Hi.. I am running Asterisk CVS-v1-0-12/22/04-05:48:41 built on a i686 running Linux (CENTOS) I do not have h323 neither OH323 configured. Accordingly I do not have h323.conf file under /etc/asterisk/sip.conf. Anyway, I wish to install oh323 in order to trunk Asterisk to Cisco 3745. What is your recommenendation,? Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Asterisk version naming convention!!
Dear list, I am running Asterisk CVS-v1-0-12 what is this called in terms of Asterisk versions convention? Is it Stable , Head, latest release !!! Excuseme if the question is too basic, but your help is appreciated. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] H323 on Asterisk@Home
Asterisk is not using any H/W at all. I am only configured it with SIP and wish to do the following scenario. Register an IP phone and a pocket PC running a SIP client with Asterisk. Then getting either to call a PSTN number. I have both Cisco 3745 and Cisco CCM. CCM currently doesn't support SIP. Therefore I will go for H323 trunking with Cisco 3745. Any ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Thompson Sent: Wednesday, January 12, 2005 4:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 on [EMAIL PROTECTED] Walid Azab wrote: Guys, I am about to install H323 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (Asterisk CVS-v1-0-12/22/04-05:48:41). I noticed that the default h323h.conf file is not set up. I also noticed that many of you here say that it is better to use Oh323. What is the best scenario here for me? Well, that depends, what does your scenario look like? What hardware, how will it be accessed(over the net)? Should I go with the already existing h323 located on (channels/h323)? or go for oh323? Choose based on features implemented, hardware known to work with each version, ability do identify, find, and download appropriate versions of software, etc. This would be a good place to start: http://www.google.com/search?q=site%3Avoip-info.org+h323+oh323 BTW, how can I get PWLib! You could type PWLib into your favorite search engine... -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.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] What's the easiest way to get * to call PSTN?
Wehave Asterisk CVS 1.0.2. I intend to connect Asterisk toCisco 3745 unless there is a better way. Asterisk is not configured with any H/W. Cisco 3745will accordingly send the call to the softswitch. PGW2200 which controls our AS5300. Walid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon EstepSent: Wednesday, January 12, 2005 3:25 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] What's the easiest way to get * to call PSTN? You have not specified what type of lines you wish to use, POTS, PRI, T1-CAS, E1, ISDN/BRI ??? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walid AzabSent: Wednesday, January 12, 2005 5:11 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] What's the easiest way to get * to call PSTN? Hi, I just want to know what is the easiest way to have Asterisk route calls to PSTN. Hope any one can help me. PS: Any solution using a Cisco device is preferable. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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 to PSTN
Thanks. Any tips on a dial plan example to route from Asterisk to CCM and vice versa? Also with H323 between * and CCM can I still use SIP phones behind Asterisk. ThanksWalid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jo?o AmaroSent: Monday, January 10, 2005 5:04 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Asterisk to PSTN -BEGIN PGP SIGNED MESSAGE-Hash: SHA1HelloYou can use H323 to connect to Cisco CallManager.Add asterisk as an h323 gateway on cisco callmanager.Then you can send receive call from asterisk.TIP: Use OH323 instead off asterisk h323 native driver.RegardsJoão AmaroWalid Azab wrote:| I have installed [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] on a PC here| and need to have it forward calls to the PSTN. We have Cisco| CallManager 3.3.4. However I found out that this version doesn't| support configuring SIP Trunks. Is there an alternative solution.| Thanks|| Walid||| --||| ___ Asterisk-Users| mailing list Asterisk-Users@lists.digium.com| http://lists.digium.com/mailman/listinfo/asterisk-users To| UNSUBSCRIBE or update options visit:| http://lists.digium.com/mailman/listinfo/asterisk-users-BEGIN PGP SIGNATURE-Version: GnuPG v1.2.4 (GNU/Linux)iD8DBQFB4plaJUm/Bor63CERAgXMAKDGJA+KXiC0FRnW7yjhJo3+YA3EMQCdEV+Ac5tmH6UTgCRW2kDr4mqNoQ4==gH7x-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] Cisco ATA 186 for PSTN dialing
Hi all.. can I configure Cisco ATA 186 to dial out to PSTN? I need a quick and easy to set up scenario to have SIP phones dial PSTN numbers. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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 enable debug
Hi.. Can someone help telling me how to enable a full debug mode and how to turn it off again. I need to see what Asterisk is doing behind the scenes. I am able to see the SIP debug events only now. But I still need to see things like voicemail to e-mail activities. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Blank Voice Mail messages
Hello, I am having a couple of problems. Any help is appreciated. 1. The voice mail messages arrive in the mailboxes but when I play them back, the IVR tells the time and date of the message but never plays it. It is as if it skips it. 2. Asterisk never seems to send the voice mail as an attachment!! Please see below my config files: Thanks Walid voicemail.conf [general] format=gsm ;already tried WAVservermail=10.150.200.5 attach=yessaycid=yes [local] ;; format: password, name, email address for attached voicemail msgs; 2000 = 1234,waz,[EMAIL PROTECTED],[EMAIL PROTECTED]2001 = 1234,StarCom,[EMAIL PROTECTED],[EMAIL PROTECTED] --- SIP.conf -- [general] port = 5060 ; Port to bind to (SIP is 5060)bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)disallow=allallow=ulawallow=alawallow=gsm context = bogon-calls ; Send SIP callers that we don't know about here [2000] type=friend ; This device takes and makes callsusername=2000 ; Username on devicesecret=123456host=dynamic ; This host is not on the same IP addr every timecontext=from-sip ; Inbound calls from this host go herenat=yes ; nat=yes if this phone is behind a NAT box or firewall mailbox=100 ; Activate the message waiting light if this ; voicemailbox has messages in it [2001] ; Duplicate of 2000, except with different auth data type=friendusername=2001secret=123456host=dynamiccontext=from-sipnat=yes ; nat=yes if this phone is behind a NAT box or firewall mailbox=101 ___ Asterisk-Users mailing list Asterisk-Users@lists.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 Demo
I am not very experienced with Asterisk yet, what will this dial plan dictate? Thanks Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes Sent: Monday, January 10, 2005 12:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Demo On Mon, 2005-01-10 at 07:51, Walid Azab wrote: Hi, I need to setup a demo for asterisk and need some help here please. The demo is connecting to Asterisk a Cisco 7970 SIP (ver. .0) and a SIP client on HP iPAQ via a wireless hotspot. I need to configure both with the same extension with a shared line like in Cisco CallManager. This way if the extension is called both iPAQ and the IP phone ring and the user gets to pick up using either. In your dialplan put something similar to: 123,1,Dial(SIP/1SIP/2) Change the exten, priority SIP number2 to suit your dialplan Your input is highly appreciated. Thanks Walid __ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.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 Demo
Ah, ok.. It is a bit clearer now.
I'll give it a try.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Monday, January 10, 2005 12:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Demo
I need to setup a demo for asterisk and need some help here please.
The demo is connecting to
Asterisk a Cisco 7970 SIP (ver. .0) and a SIP
client on HP iPAQ via a wireless hotspot. I need to configure both
with the same extension
with a shared line like in Cisco CallManager. This
way if the extension is called both iPAQ and the IP phone ring and the
user gets to pick up
using either.
You will not be able to configure both phones with the same extension.
The one that registers last will be the only one that will function, until
the second decides to register. Then the first one will fail and the second
one will work.
Give each phone its own extension and then include both extension numbers
within the Dial command (as someone already commented on).
Something like:
PHONE3=SIP/3010
PHONE4=SIP/3011
exten = 100,1,Dial(${PHONE3}${PHONE4},20)
Both phones will ring, but the first one that answers the call will get the
call.
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[Asterisk-Users] Asterisk to PSTN
I have installed [EMAIL PROTECTED] on a PC here and need to have it forward calls to the PSTN. We have Cisco CallManager 3.3.4. However I found out that this version doesn't support configuring SIP Trunks. Is there an alternative solution. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Asterisk Demo
Hi, I need to setup a demo for asterisk and need some help here please. The demo is connecting to Asterisk a Cisco 7970 SIP (ver. .0) and a SIP client on HP iPAQvia a wireless hotspot. I need to configure both with the same extension with a shared line like in Cisco CallManager. This way if the extension is called both iPAQ and the IP phone ring and the user gets to pick up using either. Your input is highly appreciated. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Asterisk SIP channel (PSTN Calls)
Hello Every one I need to enable Asterisk to route external land line calls to the PSTN. Regarding our environment, we have Cisco CallManager (3.3.4) to which IP phones are connected. E1 terminated on a couple of As 5300's which are controlled by a soft switch (Cisco PGW200 Call Control). What is the best scenario to route external calls to PSTN. Should I use SIP or just connect Asterisk to Cisco CCM. Any technical details are very much appreciated. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Aterisk@Home
Hi All, Wehave just installed [EMAIL PROTECTED]. It was straight forward as promised. However, I cannot find any guides or tutorials on how to administer this version of asterisk. We plan to install a bunch of Cisco 7960 and 7905 IP phones. I have a test phone which has already been upgraded to SIP 7. Now the box is ready but we don't know what the next step is!! Any help is appreciated. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Aterisk@Home
Yeah thanks. I got that already from the webpage. The problem is that I need a guide to help us know the correct sequence of adding phones and doing the first call. Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Soren Rathje Sent: Wednesday, December 22, 2004 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] Walid Azab wrote: Hi All, We have just installed [EMAIL PROTECTED] It was straight forward as promised. However, I cannot find any guides or tutorials on how to administer this version of asterisk. We plan to install a bunch of Cisco 7960 and 7905 IP phones. I have a test phone which has already been upgraded to SIP 7. Now the box is ready but we don't know what the next step is!! Any help is appreciated. http://your_server/maint /Soren ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Guide to Cisco 79xx
Try this e-learning tutorial. It requires macromedia flash. http://www.cisco.com/warp/public/779/largeent/avvid/products/7960/router_page.htm http://www.cisco.com/warp/public/779/largeent/avvid/products/7940/index_1020.htm Regards, Walid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul RodanSent: Wednesday, December 08, 2004 9:48 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Guide to Cisco 79xx Anybody have a guide to the Cisco 79xx phones? One that I can give the 7 or 8 ppl in my office so that they can stop asking me questions. I was going to type up a basic guide but then decided I dont want to reinvent the wheel, one of you may already have one. I tried to use Ciscos guide but its for their own protocol, a lot of options are different or rearranged. I need a basic user guide that instructs on placing calls, answering calls, putting calls on holds, warm transfer, blind transfer, activing/deactiviating do not disturb, conference calls and how to access multiple calls on hold. Like how 2 can be on hold on line 1 and if another call comes in, it goes to line 2. The only way to get back to the 2 on hold on line 1 is to hit the line 1 button. Anything anybody has would help, itd at least be a start and I can addon/enhance or even simplify/dummy down some of it. I get so many questions from ppl forgetting how to do something I think throwing a manual at them would be far superior. Any help would be greatly appreciated. We use a mix of 7940s and 7960s in SIP mode, with firmware 7-3 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Hardware
Can I start using Asterisk with a couple of SIP IP phones and Softphone software on users PCs only? I do not have any cards yet and will still have to wait until I order a card. Regards,Walid ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ZAP and IAX Trunks
Thanks Dean.. Well, about the hardware then. What do you recommend for beginning with Asterisk. I intend to use Cisco 7940s/7960s with Asterisk. Also which software is recommended to enable Soft phone on users PCs? Regards, Walid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collinsSent: Saturday, December 04, 2004 7:21 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] ZAP and IAX Trunks Hi Walid, Welcome to the list. Zap are the connections from ordinary pstn (or telco lines) to your digium hardware. IAX is an Asterisk protocol for incoming lines via IP from another asterisk PABX. Hope this helps. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walid AzabSent: Saturday, December 04, 2004 5:42 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] ZAP and IAX Trunks HelloEveryone, I have recently come across these two terms. I am new at Asterisk and do appreciate your assistance in this. Some tools such as "astGUIclient" and "Asterisk Management Portal" require that the phone system be running Zap or IAX trunks as well as SIP devices. SIP devices are understadable but what about the other two? I am planning to use Cisco 7960/7940 IP phones. Thanks Walid ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco IP Phones
What do you suggest then Brian? Thanks Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Saturday, December 04, 2004 9:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones Let me CLARIFY for those that might not get what I ment.. DO NOT RECOMMEND SCCP unless you have actually installed and used it. Its crap... SIP is what you want if you use a cisco phone with asterisk. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Saturday, December 04, 2004 1:33 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones Pfft ya right if you want half assed supported channel drivers. Use SIP. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Keith O'Brien Sent: Saturday, December 04, 2004 12:57 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones No you don't have to use SIP. You can also use the SCCP channel on * with Cisco phones. Message: 16 Date: Sat, 4 Dec 2004 12:45:53 +0200 From: Walid Azab [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco IP Phones To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco IP Phones
Guys, obviously there is an argument about SIP vs SCCP when it comes to using Cisco IP Phones with Asterisk. I am not really sure which way to go. Probably I will go with SIP now unless you guys do recommend not using it. Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Saturday, December 04, 2004 9:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones Let me CLARIFY for those that might not get what I ment.. DO NOT RECOMMEND SCCP unless you have actually installed and used it. Its crap... SIP is what you want if you use a cisco phone with asterisk. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Saturday, December 04, 2004 1:33 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones Pfft ya right if you want half assed supported channel drivers. Use SIP. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Keith O'Brien Sent: Saturday, December 04, 2004 12:57 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones No you don't have to use SIP. You can also use the SCCP channel on * with Cisco phones. Message: 16 Date: Sat, 4 Dec 2004 12:45:53 +0200 From: Walid Azab [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco IP Phones To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco IP Phones
Thanks Keith..could you please send me any useful info on SCCP usage and how I can use it with Cisco IP Phones. Walid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Keith O'BrienSent: Saturday, December 04, 2004 8:57 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones No you dont have to use SIP. You can also use the SCCP channel on * with Cisco phones. Message: 16 Date: Sat, 4 Dec 2004 12:45:53 +0200 From: "Walid Azab" [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco IP Phones To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset="us-ascii" Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAP and IAX Trunks
HelloEveryone, I have recently come across these two terms. I am new at Asterisk and do appreciate your assistance in this. Some tools such as "astGUIclient" and "Asterisk Management Portal" require that the phone system be running Zap or IAX trunks as well as SIP devices. SIP devices are understadable but what about the other two? I am planning to use Cisco 7960/7940 IP phones. Thanks Walid ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Cisco IP Phones
Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
