similar to: chan_sip notices

For bridging VOIP with PSTN Lines Which one is giving better performance SIP or IAX ? I am looking at a result without NAT in picture ? Can some body give details from experiance ? Also with single SIP/IAX channel can I use more than one call at a time ? Thanks Sandeep

Thank you very much for all feedback on Asterisk Sunday News! This is the last issue for June. This week I'll go on holiday and will be back with more news in early July. My kids are getting summer leave this week and we'll be visiting the south of England for a while. Another part of Europe that still use their own currency. If you think there's an European standard, you're

Dear All, I am totally new in this arena and I am still waiting for my installation process on freebsd to finish, but I wanted to make sure of the following: - Call routing between IP telephones can be done regardless of who made the phones? - Asterisk does support SIP V2? - it does act as SIP Proxy and Register? -- Thx MAG -------------- next part -------------- An HTML attachment was

Hi all, I'm trying to see if I can get the hint priority working with my polycom 500. So far I have 2 </reg> entries with the same sip registration, one is labeled as private, the other as shared. I have set the hint priority before anything else in my dialplan for my extensions. As it stands, I have two registrations on the phone, one has a half greyed out phone icon, the other

I checked out CVS HEAD today and tried to compile it with no luck, so then I checked out the stable version and compiled it successfully. I'm 99% sure that I'm not missing anything and that I'm following the instructions correctly (I'm no guru, but I've compiled lots of programs successfully). My question is this: is it fairly common that the CVS HEAD version won't

Well I'm stunned no one has suggested a webcast option. I mean we aren't talking a bunch of people unable to grasp the concepts of chat/voice/vision sessions with a log in/remote display capability. If you think this is an option let me know I have someone who has some software they wouldn't mind stress testing as a trial. Cheers, Dean > -----Original Message----- > From:

How do you configure an outbound proxy for Asterisk? If the sip call is not local I want everything to go to a designated sip proxy. Thanks, Chad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040913/bdd57a91/attachment.htm

I have been working with this for a wile and I have been watching the list for about a month on this subject, to no avail. I am wondering if anyone has successfully configured asterisk for clients to connect to it when the clients are behind nat. I mean successfully because I can do everything except for audio, my audio is only one way. I am asking so I can determin if I will be continuing

Hi ; Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card... I mean is it possible to direct pstn calls from astersik (extensions) to handytone line port directly and vice versa ?... Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli

We are using a Cisco router with a T1 card plugged into a PRI provided by a local telco (Allstream). This Cisco accepts calls and sends them to a couple of servers running Asterisk depending on which number was dialled. But there is a problem. When a call comes in to the Cisco from the PSTN it sends it to the Asterisk server something like this: FROM: 204XXXXXXX@<CISCO IP> TO:

One of our remote user's phone reports frequently: Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from <IP> What can I do ??? bye Ronald

I feel this is a slap in the face for those of us that have been here and I don't feel I should HAVE to pay to be certified... I think me and MANY others are about to walk out of the project over this. I have already spoken with many people that are close to the project. You're hurting US and our ability to make money. I still know the code better than most of the people that will be

I am wondering if there are any tricks getting the Cisco ATAs to do "distinctive rings" via the ALERT_INFO variable? I have seen some contradictory information in the Wiki, and I tried the example there. I then sniffed the connection between the server and the ATA and didn't see the header sent like it is "supposed" to be. If someone out there has a handle on this and

I recently got a SNOM 360 and have been trying to get the extension lights to work. I can see the subscriptions with sip show subscriptions but I don't see any notifies when a call is made. I must be missing something because I've tried looking to see if anyone else has had this problem but the only solutions I've seen have been to put hints in and I have those. Any suggestions?

We're proud to announce Astricon 2004 - the first Asterisk user's and developer's conference! * Where? Atlanta, USA * When? September 22-24, 2004 The conference is arranged in partnership with Digium.inc and the keynote speaker is Mark Spencer, lead developer of Asterisk - the Open Source PBX. Among the speakers already signed on are Ed Guy of Pulver.com, John Todd, Jeremy McNamara

Friends, Please send all offers for help *off list* to us at info@astricon.net. Do not disturb the list with offers of your services, please. I repeat: Only the Developer's Meeting will be considered for broadcast at this time. In order to enjoy the conference, you will simply have to be there. It's an IRL experience - meeting all the other Asterisk user's from around the globe,

Hello, I found, that in CVS Head, in chan_sip.c, there's some support of asterisk. I have one question -- how does it map extensions to sip user names? When my client "subscribes" to other extensions' presence, they see all users online, but it may be because of voicemail fallback. Is there a way to map extension to some sip user's presence? Any ideas are welcome.

I am going to change my Digest realm to match my DNS SVR record. I dug through the code in chan_sip.c and on line 2748 I found it hard coded <frown> : snprintf(tmp, sizeof(tmp), "Digest realm=\"asterisk\", nonce=\"%s\"", r\anddata); I'm going to change this to : snprintf(tmp, sizeof(tmp), "Digest realm=\"isdn.net\",

Astricon 2004 is the first Asterisk user's and developer's conference, to be held in Atlanta, Georgia in September. See http://www.astricon.net ** We will soon open for early bird registrations! ** To get a very low price, I recommend that you participate as a speaker. In fact, speakers doesn't pay any fee at all to participate in Astricon. **** Send us your speaker's proposal

Dear Asterisk community, sorry if I'm so stupid, but I couldn't register myself with Asterisk. I created the [sip-incoming] context in the sip.conf: [sip-incoming] type = peer username = elzhov port = 5062 ; my kphone listens port 5062 host = 127.0.0.1 Then run Asterisk, and checked peers that are known for Asterisk: *CLI> sip show peers Name/username