similar to: parameters documentation

I have followed the instructions on voip-info.org for Realtime SIP peers, but I get this notice : [Feb 22 20:05:32] NOTICE[15298]: chan_sip.c:15889 handle_request_register: Registration from '<sip:testsip at 192.168.1.150;transport=UDP>' failed for '192.168.1.105' - No matching peer found The CLI shows : [Feb 22 19:58:23] == Parsing

I have a problem with my asterisk@home installation (configured with AMP) My question is this, can you have different ports for different contexts within IAX? [Faktortel] port = 5036 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls allow=all ; Allow all codecs register => XXXXX:XXXXX@iax.faktotel.com/EXTEN

Hello, What do you think of this project www.openpbx.org ? Something like ser and openser ! Kinds Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com

I dont recall the SPA-941 playing a stutter tone in the previous firmware but it is driving me nuts, anyone know where to turn it off? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com

I live in New Brunswick Canada. The phone company is Aliant. When you set up business service here, you can go with either analog or digital lines. This isn't a T1 or ISDN. They are talking individual lines direct to handsets that they provide. They offer the digital option with even very small ( 2 - 4) number of lines. What technology could this be? Is there any way to connect such a

Hi, I know there are people with much experience in asterisk, and I want to ask if anyone had experiance with this gw http://www.eurotech-communication.com/products/voip-gateways/VoIP-32-CHANNELS-2E1-PRI-1U/ I'm having trouble getting connect with asterisk anyone has any production? regardss -- rickygm http://gnuforever.homelinux.com

We have upgraded a box from 6.2 to 6.3-RELEASE. Afterwards the box does not recognize its ST Lab I-160 serial card with Netmos 9845 Chipset. It worked flawlessly on 6.2 with puc(4) driver. >From dmesg: pci1: <simple comms, UART> at device 8.0 (no driver attached) ~# pciconf -l -v | grep -B 4 UART none2@pci1:8:0: class=0x070002 card=0x00041000 chip=0x98459710 rev=0x01 hdr=0x00

My setup is as follows: Handset -> Sipura SPA 2000 -> Asterisk -> VoicePulse and Handset -> Sipura SPA 2000 -> Asterisk -> Digium X100P -> POTS I notice when making VoicePulse calls (but *not* POTS calls through the X100P) that there is significant "popping" and "clicking" on the line. This isn't enough to interfere seriously with the call, and

I configured agi-test.agi on extension 111 when i dial into asterisk extension 111 using a IAX softphone and hangup while the AGI is playing asterisk crashes. Does anyone have any idea why this happens. -- regards Vikram (http://www.vicramresearch.com)

I have two 1-700 numbers from IAXTel. Both get registered from the same Asterisk server. I can make and receive calls on each without any difficulty. What I can't figure out how to do is route the incoming calls differently based on which 1-700 number is dialed. I must be missing something obvious. Thanks -brian -------------- next part -------------- An HTML attachment was scrubbed...

I have one SIP extension that can't logon to voicemail. The log file says -- Incorrect password '3213' for user '4035' (context=other) even though the context in voicemail.cnf says 4035 => 3213,Bill Smith Thanks! Paul Mahler mail:pmahler@signate.com phone: 650.207.9855 fax: 877.408.0105 -------------- next part -------------- An HTML attachment was

Dear All, . After installing, testing and like many other I found that Asterisk is reliable and a great Open source telephony solutions for a professional use. I would like to create a business with *, offering it as a IP PBX solution to customer, as a server, whatever with some Digium HW and SIP phone, .. etc. Asking for something that might have been asked. What are the implications for

Digium X100P / new cards are is available on ebay for $43. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309 <http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309 > Hope this helps to who want to play with X100P! Are these being sold by Digium ? I don't know ?? - SamW -------------- next part -------------- An HTML

hi, I want to complete asterisk configuration from database(MYSQL),now I come across some doubts: 1. http://voip-info.org/tiki-pagehistory.php?page=Asterisk+configuration+from+database&diff2=3 says that "Dynamic 'friends' (Asterisk v1.0.*) and the number of options supported by this 'MySQL_Friends' system is currently very limited",at the same time I find

For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the

Two cvs-head asterisk boxes with iax2 working fine (without register statements). When two calls are placed simultanously from system A -> B and the packets are sniffed on the wire, I see the two calls using two different udp packets. At the top of iax.conf I have trunk=yes and jitterbuffer=yes (at both ends). I was expecting to see both calls handled within a single udp packet, but

Hey, I'm getting an odd message in my logs, and have'nt been able to find much information on it: Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call 6010532c6fedf9be383872e07e4be70c@192.168.1.2 for seqno 102 (Request) I'm running asterisk with a Cisco 7960G If anyone know's why i'd get this.....Any help would be appreciated!

Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. Not that big of a deal I know, and the more I think about it, the more this seems a voicepulse issue. But in the off

Hi, Is Asterisk (or can it be set up as) a SIP proxy? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi all there. I am trying to set up an icecast2 server (wich is already running) to send some stream mount point to another server. I mean, my server is streaming at /radio1.mp3 mount point. We can listen it at http://myserver:8000/radio1.mp3; so, i want to know if is there any way to "clone" that strem and send it to another server, such as http://giss.tv:8000/radio1.mp3 Thanks. --