[Asterisk-Users] 7960 SIP image
Hi, If you still are in the Skinny image Settings --- Network config in that menu press **# and you will get the phone unlock. Otherwise, if you are in SIP you need to do the following: Once the telephone has booted -- Settings -- 9 Unlock config --- Enter password The default password is cisco You need to have a CCO account to download the image from Cisco site. Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more.___ Asterisk-Users mailing list Asterisk-Users@lists.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 7940G
I got similar issues, Im running P0S3-07-2-00 loadInyour tftpboot folder in your TFTP server make sure you have these files: CTLSEP000D651CF3FB.tlvSEP000D651CF3FB.cnf.xml SIP000D651CF3FB.cnf [EMAIL PROTECTED] tftpboot]# cat CTLSEP000D651CF3FB.tlvP0S3-07-2-00[EMAIL PROTECTED] tftpboot]# cat SEP000D651CF3FB.cnf.xmlDefault callManagerGroup members member priority="0" callManager ports ethernetPhonePort2000/ethernetPhonePort /ports processNodeName110.10.200.2/processNodeName /callManager /member /members /callManagerGroup loadInformation6 model="IP Phone 7910"/loadInformation6 loadInformation124 model="Addon 7914"/loadInformation124 loadInformation9 model="IP Phone 7935"/loadInformation9 loadInformation8 model="IP Phone 7940"/loadInformation8 loadInformation7 model="IP Phone 7960"P0S3-07-2-00/loadInformation7 loadInformation2 model="IP Phone 7905"/loadInformation2 loadInformation30008 model="IP Phone 7902"/loadInformation30008 loadInformation30007 model="IP Phone 7912"/loadInformation30007/Default[EMAIL PROTECTED] tftpboot]# cat SIP000D651CF3FB.cnf # SIP Configuration Generic File (start)image_version: P0S3-07-2-00. . . [output cut] Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. Learn more.___ Asterisk-Users mailing list Asterisk-Users@lists.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] 2nd try Mediatrix 1204
Hi everybody, I have setup a Mediatrix 1204, the calls worked fine, both incoming and outgoing. The problem here is the delay. When I do a call to the PSTN or receive a call from the PSTN exists a delay of 4 seconds after answer or sending the call. For OUTGOING My Dialplan for the Mediatrix box is the following, here at Mexico we use 8 digits for local calls. ([1-9]xxx|01xx||060|0xx) I have verified that inmediatly after I dial from my IP phone, the in-use light turns on in Mediatrix but the call is not pass until the 4 seconds timer expires. I have tried disabling the Dial plan butit didnthelp Form Mediatrix documentation The Timer is set to 4 seconds. It can be used to indicate that if users have not dialed a digit for 4 seconds, it is likely that they have finished dialing and the gateway can make the call. A Dial Map for this could be: [2-9]xxT FOR INCOMING The same 4 seconds delay after the call is sent to Asterisk. The problem here, is that despite we answer or not the call, once the call is sent to Mediatrix, the calling partyhear 2 ring-back tones generated by Mediatrix, then the ringback for Asterisk Once the call is passed to Asterisk and starts ringing, if we call from a cell phone,home or office the call is marked as answered and the call timer startsno matter if isanswered or not. Any ideas? I have tried sending the # at the end with no success. Thanks!__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
[Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall
Miguel, Congrats, i was testing your R2/MFC link, and I was able to made lots of calls, all of them worked fine.Thanks for setting up this link. When i hang up, there were no dead air, music on hold worked fine, when I called to a conference worked fine also, busy line Telmex recording worked also fine. Please let me know if there is anything I can help you with or if you want to test something. Thanks again! Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more.___ Asterisk-Users mailing list Asterisk-Users@lists.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] Mediatrix 1204 DialPlan and Delay
Hi everybody, I have setup a Mediatrix 1204, the calls worked fine, both incoming and outgoing. The problem here is the delay. When I do a call to the PSTN or receive a call from the PSTN exists a delay of 4 seconds after answer or sending the call. For OUTGOING My Dialplan for the Mediatrix box is the following, here at Mexico we use 8 digits for local calls. ([1-9]xxx|01xx||060|0xx) I have verified that inmediatly after I dial from my IP phone, the in-use light turns on in Mediatrix but the call is not pass until the 4 seconds timer expires. I have tried disabling the Dial plan butit didnthelp Form Mediatrix documentation The Timer is set to 4 seconds. It can be used to indicate that if users have not dialed a digit for 4 seconds, it is likely that they have finished dialing and the gateway can make the call. A Dial Map for this could be: [2-9]xxT FOR INCOMING The same 4 seconds delay after the call is sent to Asterisk. The problem here, is that despite we answer or not the call, once the call is sent to Mediatrix, the calling partyhear 2 ring-back tones generated by Mediatrix, then the ringback for Asterisk Once the call is passed to Asterisk and starts ringing, if we call from a cell phone,home or office the call is marked as answered and the call timer startsno matter if isanswered or not. Any ideas? I have tried sending the # at the end with no success. Thanks! Do you Yahoo!? Send a seasonal email greeting and help others. Do good.___ Asterisk-Users mailing list Asterisk-Users@lists.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] Callmanager 4.1 and Asterisk
You need to create a SIP trunk in CCM and in Asterisk a peer in sip.conf with the IP address of the CCM (trunk) In the trunk configuration change the transport to UDP. Enter the IP of Asterisk. And create a route pattern with gateway the SIP trunk In Asterisk in extensions.conf create the route to CCM phones. I have this setup in my lab with CCM 4.02sr1 and works so fine. If you need the sip.conf / extensions.conf and an screenshot of the route pattern and SIP trunk config just let me know! Happy holidays! Keith O'Brien [EMAIL PROTECTED] wrote: I have a similar setup. To make it easy and get the best of both worlds, have the Linux softphones (SIP or IAX) register to Asterisk. Keep the physical phones registered to CM. From there setup a dialplan on both Call Manager and Asterisk to relay calls between the two systems. For example, assign all physical phones extension 2XXX and softphones 3XXX. Have asterisk route 2XXX calls to CM via SIP and vice versa on Call Manager. Also, just so that you are aware you can register a SIP Linux softclient to Cisco Call Manager if you are running Version 4.1 --- Hello everybody, im newbie in VoIP, but find this project asterisk very interesting, i tried to install and its a great sw, i really get sorprised about all of its functions, we need to use the asterisk server in conjunction with cisco callmanager. We have a Cisco Callmanager 4.1 and the clients are softphones from cisco IPCommunicator, but all the support service of our company are linux machines, i read about callmanager uses skinny a propetary protocol and there are no softphones from linux to talk with it, so we need to install vmware to use ipcommunicator or the other solutions as i read is get the asterisk server using sip phones in the linux and windows machines and configure the call manager to talk with the asterisk server thru sip protocol, is this the real way to do that?? is there a easy way to do this?? i found this link http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration but i need to know what things to do to transfer all the extensions from de callmanager to the asterisk sw, or if only made the changes in the sip.conf as said in the link above the callmanager gets all the control?? or if i need to declare all the extensions in the asterisk?? can anybody help me?? TIA Edgar ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more.___ Asterisk-Users mailing list Asterisk-Users@lists.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] Free World Dialup and Asterisk
Hi, Julio, thanks for the tip, IAX and the incoming calls confi did the trick! FWD is up and running! THANKS! and happy holidays! Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more.___ Asterisk-Users mailing list Asterisk-Users@lists.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] Free World Dialup and Asterisk
Hi forum,
I have been fighting days and days configuring FWD and asterisk with NO success
I have the following scenario.
My sister in Spain with FWD dialup client
My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone.
Spain LAN
FWD dialup account - Internet -- 3COM router/switch --- Asterisk -- 7960
I have done some research in google with no success.
http://www.m-networks.net/home/asterisk/ast-fwd.htm
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
When I connect my FWD client in the LAN i can dial FWD numbers
ANY IDEAS OR CONF FILES WORKING WILL BE APPRECIATED
THANKS!
server*CLI sip show registryHost Username Refresh State69.90.155.70:5060 431044 160 Registered69.90.155.70:5060 421058 160 Registered
SIP.conf
register = 421058:[EMAIL PROTECTED]/103 ;Register Free World Dialupregister = 431044:[EMAIL PROTECTED]/103[fwd1]type=friendusername=431044secret=passwordfromuser=431044fromdomain=fwd.pulver.comhost=fwd.pulver.cominsecure=verycanrenvite=nonat = yesdtmfmode=inband
[fwd2]type=friendsecret=passwordusername=421058fromuser=421058fromdomain=fwd.pulver.comhost=fwd.pulver.comdtmfmode=inbandnat=yescanreinvite=no
extensions.conf
FWDUSERID1=421058FWD1USERNAME=Gonzalo GascaFWDUSERID2=431044FWD2USERNAME=Gonzalo GascaFWDPREFIX=*
[fwd1-out]exten = _8.,1,SetCallerID(${FWDUSERID2})exten = _8.,2,SetCIDName(${FWD2USERNAME})exten = _8.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],70)exten = _8.,4,Macro(fastbusy)exten = _8.,5,Hangup
[fwd2-out]exten = _7.,1,SetCallerID(${FWDUSERID1})exten = _7.,2,SetCIDName(${FWD1USERNAME})exten = _7.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],70)exten = _7.,4,Macro(fastbusy)exten = _7.,5,Hangup
My IP phone include those fwd1-fwd2-out__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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[Asterisk-Users] SIP endpoints ---- RTP stream
Hi all, I have just setup Asterisk, but the problem is that all RTP stream pass through Asterisk, I mean all call setup and voice stream pass trough Asterisk once the call is established. Is there a way thatcall setup is established, the RTP stream pass just between the SIP endpoints. Example: Works like this SIP IP phones ---Asterisk RTP stream-- SIP IP phone Asterisk SIP IP phones --RTP SIP IP phone Thanks! Do you Yahoo!? Meet the all-new My Yahoo! Try it today! ___ 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] MGCP Gateway
Any example for configuring T1 PRI with Asterisk using a Cisco 2600 series router? MGCP config? Do you Yahoo!? All your favorites on one personal page Try My Yahoo!___ 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] RE: Cisco Unity and Asterisk
Yes seems to be no reason for using Unity instead of * VM apart that Unity is windows based. is just to test SIP protocol between them = __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] Cisco Unity and Asterisk
Hi group Anyone has perform Unity SIP integration with Asterisk PBX? Thanks! Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com___ 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] Cisco Unity + Asterisk
Hi group! Anybody has implement Cisco Unity Voice Mail with Asterisk. I read the Unity can do SIP integrations Thanks! Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com___ 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] Cisco 7970 Firmware for the 7960G
Hi Michael, There are not newsthat 7970 support SIP yet, actually the most recent news from 7970´s are that they will have GigaEthernet ports. I will email the latest SIP image tomorrow. Thanks!__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] DIAL tone
Hey group! Could someone could help me configure a DIal plan in order that when i dial 9 at the beginning i receive DIAL TONE? Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone.___ 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] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual
Here is my configuration for MEdiatrix 1204, by default the 1204 strips one digit, so it is not necessary to use: To dial OUTSIDE EXTENSIONS.CONF [locales];ignorepat = 9 exten = _9,1,Dial(SIP/[EMAIL PROTECTED])exten = _9,2,Congestionexten = _9,102,Congestion To receive calls [from-pstn];Incoming calls from Mediatrix 1204, the 1204, sends an invite to [EMAIL PROTECTED] exten = ,1,Dial(SIP/100,20)exten = ,2,Voicemail(u100)exten = ,102,Voicemail(b100)exten = ,103,Hangup *** SIP.CONF ;Mediatrix Telecomm 1204[Mediatrix]type=peerhost=110.10.200.10mask=255.255.255.255context=from-sipqualify=yescanreinvite=yesdisallow=g729nat = yes In MEdiatrix 1204 use a program called Unit Manager Network a Configure the first port as extension for port 1, in option SIP. as user agent. also edit registar an dproxy SIP as the IP address of Asterisk. Works VERY GOOD with one line, although i have seen some scenarios with more than 1 line which experince problems. Do you Yahoo!? Win 1 of 4,000 free domain names from Yahoo! Enter now.___ 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] PoE injectors
Anyone knows some home-use PoE injector that works ok with Cisco 7960s? Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish.
Re: [Asterisk-Users] 7960 help
hi man, if you are trying to upgrade to the latest version, change the permissions of the file, then to the SIPmacaddress.cnf file add a line that says image version = version, copy that line from the Sipdefault.cnf file, . If the first workaround does not work, try to downgrade to version 2.3 and the do the upgrade directly from that version. I can provide you any image you need. Let me know how that works I will highly appreciate your answerJason Kawakami [EMAIL PROTECTED] wrote: I have 4 7960's that I am trying to get working but 2 of them will notupdate to the SIP image on my tftp server like the first ones did.i keep getting the error on the phone 'Defaulting CM to TFTP server' like itisn't seeing the *.bin on the server.are you supposed to have on of those for each phone? would be like cisco etal to do something like that.TIAJason Kawakami___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Yahoo! Mail is new and improved - Check it out!
Re: [Asterisk-Users] Cisco MC3810
give me a call tomorrow i could help you with your issue 52(55) 150054 54 GonzaloWayde Nie [EMAIL PROTECTED] wrote: Wayde Nie wrote: I can get a Cisco MC3810 with a mixture of FXO and FXS ports, the MC3810 comes with a built in Ethernet port and I believe it does SIP too... Will this mean that I won't need a T1 card and dedicated channel bank? ie. Asterisk connected over Ethernet with the MC3810 and the POTS lines and stations connected to the MC3810? Does it work that way? Any other limitations or gotcha's with this approach? (I'm new to this and want to confirm before I go too far down this path...)Hi Everyone,I sent a message with the above questions over this past weekend, unfortunatelyI had an email service outage and don't have the thread replies to respond toin order to maintain the discussion thread... I hope this gets threadedproperly ;) , apologies for the confusion if it does not...In any case Steve Szmidt responded: It's really kinda silly to have a great box like Asterisk and not use VoIP with it. Whenever you use a VoIP phone all you need is the network connection. That is the best way of using Asterisk.Maybe silly, but I have to do this with a stepped rollout approach... At first,I want to replicate what I have with POTS, except with separate extensions andother details but the "user interface", aka phone handsets, remains familiar...Next, I'd like to (slowly) add the "toys", IP phones, VoIP LD providers, etc... There's a good idea to have a Digium card as some Asterisk functions require a clock signal, from one of their cards.Does this mean that a digium card through the MC3810 T1 interface would providethe h/w clock whereas using Ethernet through the MC3810 10bT interface wouldrequire a less accurate s/w clock?Does anyone know if the MC3810 FXO/FXS ports are accessible through the built inEthernet 10bT port (inferior s/w clock or not) or do you need to go in throughthe T1 interface? Has anyone actually done this? (I'm not really prepared to bea pioneer here ;)Grateful for any insights! Thanks,--Wayde Nie.___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage!
[Asterisk-Users] Mediatrix 1204
Someone have the MIB for MEdiatrix 1204 version 2.4.10.68? thanks -- Almada Tres SA de CV Mitel Networks Eng. Gonzalo Gasca Meza Service Engineer 52+(55)53730570 Mexico City, Mexico ___ 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
