similar to: Command to light MWI on 7940 /7960

Hello: Has anyone on this list had to configure hairpinning on a Cisco gateway running IOS 12.2 or 12.3 and using a PRI for connectivity to the PSTN? If so could you tell me how it is done? I'm told this is the source of my call transfer problems and yet I cannot find clear instructions for how the configuration is done. Thanks,Steve -- ISC Network Engineering The University of

Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java

Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW:

Hello to all Does someone knows how to force the Cisco IP phones (7940 and 7960) to reboot weekly or monthly? I think this would be useful because sometimes we change the configuration settings in the TFTP, but the phone just check the TFTP when he restarts... Thanks Jo?o

I can part a call (dial #700 it is parked on 701) but if I dial 701 I am told it is not a valid extension? I have include => parkedcalls in my local extension context. I have Ttr on all extensions and the incoming pots line. It parks, plays MOH but I can't retrieve it. --john -------------- next part -------------- An HTML attachment was scrubbed... URL:

I was looking for something to connect a couple of POTS handsets to my asterisk server and found this on ebay http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&category=162&item=5162868118 &rd=1 The documentation says that it does SIP - therefore will it work in an asterisk environment. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus

Could anyone shed any light on how SER and/or Asterisk (stable branch) has held up for them in that last while? Are you using SER and/or * in a production environment? Do you ever restart the software or reboot the system? How many users are utilizing the system? How many calls per day/concurrently? I read some uptimes and such on the mailing list from long ago, so I was wondering what some more

Since I can't seem to get anything figured out for the Comedian system, are there any other systems out there that we can hook asterisk into? Sherwood McGowan ViaTalk Level 2 Support VOIP System Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050912/32ae9d25/attachment.htm

OK all. I need some help. Looking to deploy asterisk servers and want to get a recommendation on what server to buy. I love Dell's, but from what I see on the list they seem to have some issues. I would like to stay with one brand and need systems for small offices (20 users), medium (50 users) and large (100 users) systems. Thanks for the help. Keith

It seems on my phone, which is hooked up to a large pbx network powered by an asterisk server, that it will randomly start ringing with a callerid# of 2013 which is its username for that phone. I have looked and been watching on the asterisk command line with the -vvvvvvr switch and nothing has been seen that indicates a reason for this random ringing. This leads me to think that this trouble

We're running Asterisk 1.6.1.17 for our campus voicemail server and Juniper M120s as our SBC. Unanswered calls, which arrive via the SBC, are diverted to voicemail using a 302 redirect when the called party doesn't answer. In this case the caller is able to hear the greetings and begin to leave a message only to have Asterisk terminate the call mid-recording. We're uncertain why

Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and

I'm looking at ordering a server from HP. I checked around on Google and found in the Wiki that the ProLiant DL380 is supposed to be known to work with *. I'm going to get a price quote on the following setup: HP ProLiant DL380 G4 Server w/ the following options: Intel Xeon 3.20GHz/1MB 2GB REG PC2-3200 (2 X 1GB) HP ProLiant Battery Backed Write Cache Enabler for SA6i RAID 1 drive set HP

Hi all, I have some probleem detecting DTMF send by a GSM phone, I'm using SIP with ulaw. do you know what are the options to improve the detection ? I'm using asterisk 1.05, is the CVS HEAD version had some improvement about DTMF detection? Florian.

Is there a protocol I'm supposed to use here? It seems that people are asking 100 questions a day and SOMEONE is helping them, and I've posted this three times and not even an "I Don't Know". My third repost: Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has

Some of you may remember back in May the thread on using Asterisk as a voicemail server for a Cisco Callmanager system. My own Callmanager system is integrated into an Asterisk server for voicemail (and other things). Back in May I was using H323 for integration, but since I've upgraded to CCM 4.1 I have switched over to SIP. The integration with H323 required using Call forwarding to send

Hello: We offer several options for accessing Comedian mail via Asterisk and IP phones. We'd like to add one more. We would like to be able to save a message on the Asterisk server in such a way that it can be retrieved using the telephone set and Comedian mail as it is today and/or via an IMAP client. If accessed via am IMAP client we would like to use an "externotify script" to

Hi all, I'm having sudden MWI problems. Everything else on the phone works fine though. I have three Cisco 7940s. Asterisk server is behind a firewall running NAT. (192.168.1.202/24) Phone #1 - On the same subnet 192.168.1.250. Everything works great. Phone #2 - On a different subnet, 192.168.2.0/24. Everything works fine except the MWI. It never comes on. This is over an IPSEC VPN, but

Hello. I am using a snom 320 running 5.3.6 with Asterisk 1.2.4. In the sip.conf entry, I have mailbox=1234@default and vmexten=*98. The light on the snom 320 turns on when I have voicemail and the retrieve button dials the correct extensions. However, the light turns off immediately after making the call to voicemail, even if I do not check the voicemail. Any idea on how to get this to behave

Hello, I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image