similar to: sip issue with one way audio

All, Since I have now (at least partially) got my 7961G phones working with Asterisk, I have temporarily moved on to try to get the expansion modules working. There doesn't seem to be much in the way of documentation here either. Does anyone have this combination working (or any 79X1) here? My goal is ultimately to do the monitoring approach. I have Google'd around, but come up

All, I have run into some CallerID issues. It seems to have happened as a result of just moving my config from 1.2.12 to 1.4.4 (although I am not sure of this). Therefore I am sure its just a misconfiguration somewhere, I just don't know where. I have throughout the office either Cisco 7961G or Polycomm Soundpoint SIP 430 IP phones. The problem is with CallerID showing up in some

Hi, Has anyone here ever used a Polycom IP 4000 Soundstation SIP Conference Phone with asterisk? If so, how well does it work and how does it sound?

hi, I am facing some issues while using MixMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(number)}-${TIMESTAMP}-${UNIQUEID}.gsm,b) in this extensions TIMESTAMP is not working in Asterisk 1.4. can any help me why TIMESTAMP is not working in Asterisk 1.4. regards, Asif

I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server. My PBX has experienced several core dumps the last couple of days and I am not sure if this is what's causing it, but it always seems to happen when a particular extension on a grandstream phone uses ChanSpy SIP group. I have not been able to locate where the core dump file is being saved. I can't find it in my

All, I've a question on A*k hardware. I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz) with 512mb RAM. I'm supporting 60 users (Cisco 7940s each + Xlite PCs). Call loads are low, max of about 10 simultaneous SIP/IAX calls. CPU for A*k rarely goes above 2% as I can tell. Its IP only, no E1/T1 cards present. However, I get complaints of bad voice quality,

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no

All, I am having a lot of trouble with the Cisco 7961G phones. I have managed to get them up and running with Asterisk to the point where I can get incoming calls and make outgoing calls. The problem is when I make outgoing calls or extension to extension calls, the calls die after 20 seconds. I have google'd around and came up with little that is of help. The firmware version I am

I used to have fonality and I could press * when I got to someones voice mail to go back to the menu. I assume I add that to the dialplan but how? Thanks BTW I went back to 1.2.12 and transfer works and DTMF works and it seems to be much better for now. Thanks for you help Jason

I put my voicemail groups into different contexts so that I can use Dial by name and escape. I had set ext 500 as exten => 500,1,VoiceMailMain(${CALLERID(number)}@default|s) but now that the contexts are different. this does not work #1 how do I have everyone use an ext to get the voicemail regardless of context. #2 can I get the mail buttons to work on my polycom 501s and swissphones #3

I have 1.2.12.1 Voicepulse using IAX I get about 30-40% issues with not having the DTMF tones work. I have 3 questions #1. Voicepulse says they are sending them, Is there some setting I can adjust to make sure my end is working? #2. I have set the Dialplan to play a sound "Operator" then go to a SIP phone. I would like it to write Caller ID Time .... to a file I can read and find

I had to switch quickly to 1.4.9 from1.2.4 and now I can only get 2 digits into the dialplan. error -- Invalid extension '81' in context 'impact' on SIP/207.174.111.34-b77167f8 I pressed 8107 and ideas my dial plan is (part of it) [impact] exten=>s,1,Answer() exten=>s,n,Set(CALLERID(name)=Impact - ${CALLERID(number)}) exten=>s,n,Background(IMPACT)

I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct DTMF tones 25% of the time. I have to call several times to enter an extension. I have a router and a packet shaper and some other stuff. Anyone have any other ideas why this might happen. I do not have any Zap channels but I am running CentOS4. I also do not have any cards installed. Thanks

Hello, This is my first asterisk installation, and having read up on the documentation, and trying several tutorials i'm unable to get my outbound route working. I'm certain it's an issue with my configuration and my inexperience with asterisk. So i have my POTS phone connected to my digium card, and when i make a call, I receive the "cannot be completed as dialed" message.

Hola!! tengo un data.frame donde cada fila corresponde a un año y cada columna a un mes (De enero a diciembre) > head(valT) V2 V3 V4 V5 V6 V7 V8 V9 V10 V11 V12 V13 1941 18.0 16.3 15.2 10.1 8.1 8.3 8.8 9.2 7.9 12.2 11.9 14.6 1942 17.2 15.9 13.6 11.6 8.7 6.2 6.4 7.2 9.7 12.0 14.1 16.7 1943 17.6 17.3 13.5 12.5 10.5 7.0 8.2 7.9 -999.9 -999.9

Hi, i have some questions when using Btrfs on multi-devices: 1. a large file will always be stored wholely on a device or it may spread on some devices/partitions? Btrfs has option to specify it explicitly? 2. suppose i have a directory tree like that: Dir_1  |--> file_1A  |--> file_1B  |--> Dir_2          |--> file_2C          |--> file_2D If Dir_2, file_2C  on a failed device,

Hi, How would I go about troubleshooting this: [Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc32ed at 192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaaec88 at 192.168.1.95 for seqno 101

Dear all I'm working on the implementation of the MDCT for the processor ARM-Cortex-M4 I'm trying to replicate the behavior of the MDCT for several overlapped values however I realized that current implementation of the MDCT is very close to the theory only in case in which we have and overlap exactly equal to N/2 where N is the size of the input vector as is shown in the examples for

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!

Running * with default config files except for sip.conf. Any call made is dropped 5 seconds after connection, with the following messages: Mar 17 16:37:41 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call 6C94C1B1-77C4-11D8-91FB- 000A95DA04DA@192.168.1.152 for seqno 48221 (Response) == Spawn extension (default, s, 5) exited non-zero on 'SIP/2000-6bd7' Mar