similar to: Sound Quality.

I'm trying to pick up a ringing SIP phone (203) across the office with exten => *9,1,Pickup(783743) where 783743 is the local part of the number that our ISDN works on. I tried all of these first: exten => *9,1,Pickup(203) exten => *9,1,Pickup(SIP/203) exten => *9,1,Pickup(203@internal) and got a "declined" message back from my phone (snom 300), so I then

We are getting the following on 1.8.3 and 1.8.4-rc2, HELP! Why is Asterisk unable to transcode to/from ulaw and g722? [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw,

Hi Guys, I have a the following issue when I ma trying to place a call through my voip provider, I am using an asterisk 1.2.21.1, do you have an idea what could fix this issue (as you can see when the other party answered, the call get dropped because of probably sip incompatibility) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 256, while native formats

With asterisk 1.8.7.0 has been running ok for months. Now, this morning, it's crashing. I can restart it, but it crashes after 10+ minutes. It dies like this -- Executing [s at macro-stdexten:2] Dial("SIP/teliax-00000019", "SIP/176,18,rtT") in new stack == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP

Does anyone know of any startups using Asterisk? What about established companies? Ones that are hiring would be nice.... :) Doug.

Hi list! I have 4 numbers on my Asterisk 1.8. 3 work perfectly, the 4.th not. I'm not sure, when it finish to work, since a month ago it runs without any problem... Well, if I will be called on this number I can't hear anything and in Asterisk I see these: [Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-000001d1 setting write format

Starting with Asterisk 13.1 we are seeing this WARNING messages a lot in our logs and console: WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type frames with SIP write) We found that line in function "sip_write" inside "chan_sip.c". In our previous version (11.2.1) we did not see those messages being printed (same verbosity level). We compared

Hi, I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used?

Hi I've problem with incoming call quality to GSM gateway connected to beronet card (BN8S0), -----> [ GSM Gateway ] -------> [ BN8S0 ] ==== asterisk Port connected to GSM gatway is in TE mode , gateway is in NT mode , When I dialin to cellphone numer , call goes to 'from-eragsm' context, to Echo application. [from-eragsm] exten => 700,1,Goto(600,1) exten

Hi all! Quite a mystery. The following happened when I was on holiday, and no one else has changed any configs of either Asterisk or the Cisco's in the building... The situation: Incoming works fine on all phones. Outgoing only works from non-Cisco phones. When calling from a Cisco phone to an external phone, all the Cisco-user hears is a ticking crackle and after about a minute the phone

Does anybody know of any IP phones (ideally SIP based) that have interfaces to plug into a pro audio system (eg for phone interviews). Something can probably be hacked up with a headset connector or the 1/8" jacks on a 7970 but I'm wondering if there's something better out there. Thanks, Julien

Hallo Waffenmeister! Ralf Hildebrandt <Ralf.Hildebrandt at charite.de> wrote: > > Apr 7 14:07:52 delta postfix/qmgr[19078]: 1D8921B31260: from=<anmeyer at anup.de>, size=1492149, nrcpt=1 (queue active) > > Apr 7 14:07:53 delta postfix/pipe[19091]: 1D8921B31260: to=<miles at anup.de>, relay=dovecot, delay=2542, delays=2542/0.01/0/0.29, dsn=4.3.0, status=deferred

http://www.x100p.com/products_2.htm Anyone ever use this box? How's it compare with the Iaxy? I'd like to buy one or the other.. The Iaxy is appealing because to me, it seems less "no name", but this one says that it supports using hostnames, whereas apparently the iaxy only supports IP addresses?? That's appealing to the dynamic DNS guy in me! :-) Any

Hello Everyone, We have recently installed a pair of Trixbox servers in for a client of our. They have two locations, with one server each. The servers terminate 3 standard POTS lines into a Sangoma A200D card. The servers are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, Dual Core Xenon Processors). We are using Trixbox 2.2, and G729 all around. Each site has two (2) 512k/512k ADSL

Hi I'm looking for some help with Vicidial, If you have experience with it and could help with some consulting please contact me off list. Cheers, Joel Hill Asterisk IT jhill@asteriskit.com.au

Good afternoon list. I'm experiencing a problem with my SIP channel's. When I have an external connection for one of my SIP carrier's, I can listen to the client and the client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. In the console of Asterisk generates the

Good day! Have a weird problem with HT-286 and Conference room. I use Asterisk CVS-HEAD-06/04/04. Here it is: When HT-286 get into the conference room first and nobody in that room everything seems ok (with any codec HT286 allowed), but when HT-286 get into conference room when somebody already there, have got different HT behavior: 1. When HT use GSM codec => it connects to conference room,

Hi everyone, I have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems that I can solve an error which gets my incoming calls working but that in turns means my outgoing calls don't work. - Strange. Anyhow I was getting an error: Process_sdp: No compatible codecs! And from

I patched samba to always return ACCESS_GRANTED for testing. So I came to this: IASSAM.LOG [556] 23:23:53:671: Inserting attribute msNPAllowDialin. [556] 23:23:53:671: Successfully retrieved per-user attributes. Dialin now "only" fails with "Dialin not allowed for user", but I'm not able to set it in UserMgr. Is it difficult to map this attribute? Daniel

Hi All, I'm looking for a good voip hardphone that has a decent set of the "regular" features (conference, 2 lines, etc) thats reliable, has decent quality, and isn't too pricey. Does anyone have any suggestions? Thanks in advance. Derek -- Derek Fedel