similar to: res_motv: Request for Comment

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, 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

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 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 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

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

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 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

On 4/6/20 9:33 AM, mark wrote: > On 2020-04-06 03:57, J Martin Rushton via CentOS wrote: >> If you have any choice in the matter I would suggest a read of >> https://en.wikipedia.org/wiki/Zoom_Video_Communications#Criticism >> before using or installing Zoom. >> > I can argue against zoom, and have, but organizations I belong to or > join, like this past

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

Hi all, Is it possible to use a nfs share to serve as a glusterfs server process?? For example: I have a host that mounts a nfs share from a solaris server under /mnt. Can I use this /mnt directory for a glusterfs server?? Thanks. -- CL Martinez carlopmart {at} gmail {d0t} com

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

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

I''m trying to test some code that has validates each and I''ve got a very strange failure Mock ''Book_1027'' expected :store_with_privacy? with (#<Clip:0x1a99b96 @name="Clip_1025">) but received it with (#<Clip:0x1a99b96 @name="Clip_1025">) The Spec it "should check that a book can save a clip" do @user =

Another trivial question: Is there a "place" where all the parameters are documented ? In example (my example!) I would like to know the meaning of a lot of parameter that can be used in sip.conf, A lot of these keywords are intuitive keywords (i.e. NAT=YES/NO;PORT=5060; context=xxxxx) but other are not (at least for me) i.e.: type = peer, friend insecure=very host=dynamic and so on.

On 25/05/15 09.51, Damien Miller wrote: > I'm not sure it should be part of the banner exchange, though there is > no other trivial way to do it and maintain backwards compatibility. Even if backwards compatibility wasn't a requirement, I don't see any better way it could be done. > I don't much like it because it reveals host identity information > in the clear. So