search for: outband

I'm using asterisk as a sip client with a sip proxy server... I've made the pertinent extensions and I've configured the sip.conf correctly or I think so.. I'm using x-lite as a client and when I ring to a public telephone through proxy, the arriving sound it's perfect but the sound I send is very bad, they hear me like a robot and distorted. Anyone know what's the

Hello, I am setting up 2 ISDN 4 linux cards and have had great success now that I have got over the initial problems with : and / characters. The only problem I am experiencing now is the sending of DTMF tones over the line to a remote IVR system. If I dial SIP (Cisco 7905 and 7940) to a number over the line, no DTMF tones are heard. I dialed my own home phone and tried it, no matter which

Hello, I am trying to make the following... Can someone tell me if it is possible? Is someone willing to do it from an asterisk@home box? 1. I send an http request to asterisk@home box. Ex: http://asterisk@home/call.php?phone=0033102030405&code=12345 2. Application will call phone number 0033102030405 (using a sip provider) 3. Application will play a pre-recorded voice prompt 4. Application

Hi all, I'm setting up my first * install and have it peering with another * machine using IAX across the internet which provides our pstn gateway. So far I have the IAX "friend" set up correctly but when I make a test call from an external phone, I get: WARNING[5126]: File chan_iax.c, Line 648 (get_timelen): Don't know how to calculate timelen on 8 packets I have set up a

Hello list i have an issue with my dahdi_channels.conf in span 1 when i use it like below i can do my outband calls without issue ; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel => 17-31 context = default group = 63 but when i add in channel 1-15 like: channel => 1-15,17-31 i receive all the time thi...

hi, i have connected a SNOM 200 to the asterisk. here are my settings, Codecs ------- Default codec - g.711u/g.711a Packet size - 20ms Negotiation - Interoperable Type - 160 DTMF ---- Inband - negotiate Outband - negotiate Payload Type - 101 when a call comes to the SNOM or when making an outdial, following warning messages are coming on asteisk, WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_process): U...

...se one of the IVR options, i notice in the /log/asterisk/messages this line: WARNING[43028]: Discarding too big frame of size 1280 These are the combinations i've tried: (I'm testing now on Snom only since we are changing all our phones from GS to Snom for quality issues) 1) Phones with Outband=on, dtmf payload=101 and dtmfmode=rfc2833 in sip.conf. 2) Phones with Outband=on, dtmf payload=101 and dtmfmode=info in sip.conf 3) I've also tried to switch to inband (setting audio codec as g711u) I'm using the 1.0 first stable version of * now, downloaded yesterday night to see if thi...

...ck to play again with my dust collecting 2000W. Does anybody got DTMF to work? My sip.conf looks like this: [400] type=friend context=from-sip username=400 secret=verysecret disallow=all allow=g729 dtmfmode=rfc2833 host=dynamic nat=yes qualify=300 canreinvite=no My phone is set to use DTMF 'outband' any ideas? Dominique -- taridium.communications dominique kull, partner the old lodge, london sw6 6ee uk t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: dk@taridium.com

Hello asterisk-users, i got a simple read configured as an extensions command. it does not detect the dtmf digits all god 0123456789 gives something like 0223455789 what could this be the issue ? i'm using g711u law with inband. i tried to use outband but i got hardware like FXO's that don't support outband -- Best regards, Danny mailto:dannyz@belgonet.com belGOnet.com a Euro-pictures division - internet solutions place princesse elisabeth 9/11 - 1030 Brussels - Belgium Tel : +32-(0)2-215.67.65...

...oCodes to allow DTMF tones to be sent after an outbound call is connected(phone banking, long distance provider etc...) while still allow the client devices(phones) to access Asterisk voicemail. It seems I can either have the phones use inband DTMF and work with the Audiocodes PSTN's or outband and work with Asterisk, but not both? Any help/thoughts/experiences would be appreciated... -joe

...ere are some features I got from the brochure: 1. MGCP, H.323 (v4) and SIP support 2. Selectable, multiple codes (g711/g723/g729A) per channel 3. G.168/165-compliant adaptive echo cancellation 4. Echo canceller jitter buffer, VAD and CNG 5. complete voice band signalling support 6. provides inband/outband DTMF generation/detection 7. provides call progress tones 8. web management interface 9. LAN (10/100) and WAN RJ-45 ports What should I be looking for when I'm testing the unit? Anyone can offer some hints as to what I need to look out for? I know voice quality should be something to look at...

...at when I place a call to outside, via E1 trunk, sometimes I get some DTMF tones and I'm sure nobody hit any key. Seems like Asterisk is misinterpreting some voice frequencies as DTMF tones and is regenerating it. I think it is related to the INFO method, as Asterisk and/or PAP2 have to send it outband and the other side will generate the TONE. Is that right? Anyone experienced something like this, and have resolved it?? Ok, the second problem is that some DTMF tones I send from my phone (Connected to the PAP2) are not being interpreted by the other side of the call (generally bank systems). I...

i need your help regarding some issue related to the outband calls i have installed asterisk 1.4.32 with dahdi and i have 1 card diguim with 2 ports when i try to call my phone number all time i receive message busy number this error just with g1. with g2 there is no problem i can call without issue can anyone see the CLI and tell me what is the problem...

Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fine if I call the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" <205> host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729

Hello All, I'm trying to get my SNOM 200 to work using the GSM codec. The problem is it wont' pass dtmf digits. I have it dtmf outband on, inband off. In my sip.conf I have dtmfmode=rfc2833. Whenever I press a digit on the snom phone I get "invalid gsm data" so I think it's a configuration problem in the snom, anyone have a working configuration using gsm? Thanks -Matt -------------- next part -------------- An HTML...

...fc2833 On the telephone web interface: Zyxel config SIP/outbound Proxy config Proxy IP:10.10.3.5 Proxy port = 5060 SIP Config SIP URI: sip: 898 @ 10.10.3.5 : 5060 Expire time: 300 Registrar username: 895 Registrar password: ---mypassword--- DSP setting Default Voice codec G.729 8k DTMF relay: outband -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/

...a welltech fxo 3802 also to * the extensions are configure so that i can dial from the lanphone to the fxo. although once on the FXO and having the dialtone, no of my dtmf dialing is being processed on the FXO. It keeps giving me the dialtone constatly. Everything is configured as outband DTMF (we tried also doing inband but it was doing the same) We tried to call the echo test application to check if the # would close the connection, and it does. So dtmf from lanphone to * seems to work OK. We also tried to configure a extension like dial(SIP/FXO,50,D(0022156765))...

Hi, I flashed my ATCom AT320 phone (PA1888S based) with IAX firmware instead of SIP but now call transfer doesn't work neither using phone buttons nor using Asterisk features. I heard that it can be a real problem. Any help? Mimmus

...idziane *Dodatkowe wymagania (mile widziane):* - umiej?tno?? programowania w PERL i JAVA - znajomo?? Asterisk / SER / OPENSER - umiej?tno?? konfigurowania bramek i ruter?w Cisco, Patton, lub innych - znajomo?? protoko??w SIP lub STUN - znajomo?? problematyki NAT I funkcjonowania outband Proxy - znajomo?? narz?dzi monitoruj?cych jak Cactus, Nagios, MRTG - znajomo?? narz?dzi jak DRBD, Hearthbeat - dyspozycyjno?? Wszystkich zainteresowanych prosimy o przes?anie ?yciorysu oraz listu motywacyjnego (zawieraj?cego zgod? na przetwarzanie danych osobowych) *do 10 wrze?nia br. **...

I'm having a strange problem with transfers on IAX phones. I have two IAX phones behind my firewall that are extensions from my office phone system. Both phones can receive calls, but only one of the extensions can do blind transfers by pressing the # key. I have a similar problem at the office. Some of the phones can transfer calls, some of them can't. And my Zap lines can always