similar to: IAX and call transfer

Hi, I have a few of Atcom AT320 phones. I can choose if install SIP or IAX2 firmware. Any suggestion? Mimmus

I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I

Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Does anyone have any recommendations on an IAX Desktop Telephone or ATA Device. I currently have 2 of the SIPURA-841's on my local network and now I am wanting to try an IAX Device at my remote office since I think that it would be easier to configure through various routers than a SIP Device. I just started to look at the Digium IAXy Single FXS

Hi all, The firmware I used is pa168s_iax2_us_151011.bin. My problem is the handset dial before I finished key in all the numbers, no matter how fast I managed to press the keys. It appeared it always dialed immediately, for example "011862", when I actually ment to dial 0118620xxxxxxxx. Thus left the remaining numbers "0xxxxxxxx" unsent. The handset had its dial plan

Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) --

Hi, I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (only one until now!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR

Hi all, I have problem using DTMF on Snom Phones (300, 320 and 360) I read they use in preference out-of-band DTMF , and if the remote system does not support it they default back to inband. I would like to use DTMF as out of band , and I defined dtmfmode=rfc2833 in the peer configuration. Nope, I am no able to access any ouside services using DTMF; Another kind of phones, ATCOM AT320, can be

Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? Thanks -- Domenico Viggiani

Hello I'm thinking of selling an Asterisk server based on Atcom's IP02 solid-state unit with one FXO and one FXS ports: http://atcom.cn/En_products_IP02.htm By default, this unit based on a 400MHz Blackfin 532 chip only has 64MB RAM and 256MB of NAND flash. Those can be increased to 128MB and 1GB, respectively. Do you think I can install Linux + Asterisk + LAMP (replacing MySQL with

Hi, I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGI script "dialparties.agi" to find the extension to call. I'd like to drop this script: does anyone can explain me what is its main job? Thanks -- Domenico Viggiani

Hi, I'm experiencing brief pauses during my calls: 0.5-1.0 sec of silence if call continues for more than a few minutes. I'm sure that problem is in the phone (a cheap ATCOM AT-320 with latest SIP firmware) but I'd like to diagnose better. During a little test, it seems that there is no problem with IAX2 firmware (but there are others... I'm not able to transfer and pickup

I've gotten nothing from http://bbs.atcom.cn on this so far. Perhaps someone on the list has experience with this. Is there a way to get MWI support for PA168V-based ATAs? Apparently some IP phones based on the PA168V chip has this support already (Atcom AT-320 for example) by configuring Asterisk with 'mailboxdetails=yes' in iax.conf. On my ATA, however, it does nothing. Any

Where can I investigate the origin of brief silences during calls from/to my SIP phone? Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything. Thanks Mimmus

Hi, I'm seeing a lot of 'h' as destination numbers in my CDR logs. Some time ago I solved this problem but now I'm not able to remember anymore. Something related to match-all extension? Any help? Thanks Mimmus

Hi all, If I activate recording for an extension everything is OK. but If I activate call recording on demand i am non able to start recording In principle I should have to press *1, as indictaed in features.conf (I am using almost last asterisk code, updated 2 days ago from svn, version SVN-branch-1.2-r39379M ) Actually it produce no effect at all I am using FreePBX interface, and I saw

Hello According to this article, this nice little unit can only use the PSTN port for outgoing calls (ie. as a backup in case the connection to the VoIP provider stops working), but not incoming calls: http://tinyurl.com/mwjmo8 Can someone confirm that Atcom made this strange decision, and that there's no work-around? Thank you.

Anyone here used these phones? I'm getting more and more frustrated by todays modern crop of routers with their so-called SIP ALGs which are invariably broken, or routers with built-in ATAs which block internal SIP phones from working, so looking to use IAX for some end-users. I already support it for people who want to use (eg) Zoiper and use IAX a lot to plumb boxes together, but never

Hello For customers who need a small IP PBX to handle up to four ISDN lines (in France, so I guess that means EuroISDN) instead of a PC + Asterisk and an ISDN gateway box, has someone already played with the Atcom IP-4B? www.atcom.cn/IP-BRIM.html Any feedback appreciated.

Hi all. We have Snom phones which do have a defined key in order to drop incoming call WITHOUT answering. Pressing that key, a "SIP/2.0 486 Busy Here" message is sent back. We have other phones (I.E. DECT Siemens C450IP, or ATCOM 320 or other) which DO NOT have any key to do that (or the key does not work, as is with Siemens C450 IP ): you have to answer and immediatly after hangup the