similar to: Cisco gateways and hairpinning

Hi, I am using real-time, oh-0.7.2, G729 Calling from (SIP)UA through asterisk towards h323 devices or the other way round, I get only one-way audio. Called party can only talk, caller can only listen. Calling SIP to SIP is ok. All devices are on official IP addresses. (no NAT) Regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail

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:

We have several agents on queues, and want to indicate to them that they are logged in or logged out. We have tried several different ways, from changing the screen to presenting different service menus, but cannot get anything to be "in their face" (their words, not mine). One of our team has suggested, as the agents do not have voicemail, is to use the MWI on the 7940 phones to

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

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

We have one Swissvoice IP10S running SIP firmware. Recently, I've been getting these messages: Mar 1 13:59:44 NOTICE[20933]: frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Any clues off the bat? I'm still researching other stuff.. Thanks, Matthew

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

Hi list! Sorry for this slightly off-topic message but does anybody know if the standard for ISDN BRI is the same in Spain as it is in the rest of Europe (or the Netherlands). Will a standard HFC-S card work?

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 Fellow *-ers, First, you guys are fantastic. Keep fighting the good fight. Second, it sounds like comments in the code are coming, which sounds welcome by all, even those of us who couldn't code their way out of a papersack, but who need to read the source. Last, I might be traveling to Europe (from US) & want to tow along hardware & haven't done this before & was

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

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:

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.

I have the following configuration and am obviously missing something small that is causing * not to work as expected. I have the following defined in sip.conf [ccme-in] type=peer host=10.0.9.1 context=devel_in disallow=all allow=alaw nat=no canreinvite=yes qualify=yes and [devel_in] is defined in extentions.conf However when I try to call via the dial peer I have configured on the cisco

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use