similar to: *-1.0.7 DTFM => Not working

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

We have some guys working in the US who can't always dial back to our company in Europe easily (lots of clients require authorization to make international calls), so I set up the following: - ipkall.com number links to a FWD number - office Asterisk box registers with FWD Then I programmed Asterisk to accept office extension number using DTFM tones. This works OK. Then I programmed

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Hello, I've got a mixture of SPIP 300 and 500 phones in production for various clients. I've got the XML settings configured for local conferencing, but I'm not seeing the expected behavior from the phone when I attempt to conference two calls together. According to the manual, while talking to the first party, you simply hit Conference, dial the second party and then Conference

Hi list, I'm having trouble with DTFM tones detection. Usually, some tones are being received duplicated in Asterisk, some not. As you can imagine, that's a very big problem involving IVR menu options, Meetme conferences protected with passwords, and so on. We are currently using DAHDi 2.2.0.2, module wct4xxp, which is managing a Digium TE220B card, with a hardware echo canceller

Hello, I am sorry that I post questios regarding Broadvoice here, but unfortunalelly their support is very very bad. The simply do not answer to any emails or telephones. Last week something happened to their system. I was not able to receive incommming calls etc. Now it is back, but the voice quality is terribe and the DTMF is not working.(Is the inbound mode the correct one?) Does anybody knows

Hi All, I am trying to compile meetme2 in my asterisk box and getting the following compilaton error. Please help me to sort it out. cc -fPIC -c -o app_dial.o app_dial.c In file included from app_dial.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function)

Abwesenheitsnotiz: [Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6Hi All I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb ram, with g729 for i686 , (fedora 1). my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen otherparty realtime voice , but other party geting sip party's voice 1 sec later (not

Hi! I'd like to ask whether someone did test Opus in real-world VOIP (SIP). Did someone e.g. some characterization about sending faxes or DTFM through Opus? Does it work and if yes for which bitrates? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20130518/907c5cbf/attachment.htm

[Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid extension '11', but no rule 'i' in context ...[snip]... When testing IVR and pressing "1" from my Grandstream SIP-phone, the above message is printed on the Asterisk CLI. How come Asterisk receives my "1" as "11" ?? Settings in my SIP-phone are : Send DTFM : via RTP(rfc2833) &

Hi Gang Not a specific Asterisk Question. But I wonder, if the called party replies with 183 + SDP indicating support for telephony-event. Should the caller be able to send DTFM Tones? Swiss Railways uses an IVR that kicks in before the call is answered. So far I have found no SIP Phone which would allow sending RFC4733 during the early audio phase (so I cannot test if Asterisk would forward

I'm having a very annoying problem with access my asterisk system from work. Our phone system here only produces very very short DTMF tones. The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE => CALLUK.COM 0870 => IAX Inbound The

Hello,Maybe I don't understand what DTMF in ASCII means but I can't make my record stop using this syntax in a PHP agi script :fwrite(STDOUT, "RECORD FILE /var/lib/asterisk/ENR/jeanpaul wav '#' 15000 BEEP s=3000\n");The php syntax isn't a problem because I really start recording, I have a beep, the record can't long more than 15sec and after 3sec of silence my

I'm testing Teliax tall free number line and I'm experiencing long delay about 1sec. during conversation. When I call myself over FWD the response is normal no delay or cut messages. When I call my number over FWD the is a long delay, welcome message usually cuts off few first words and during conversation my voice arrives after about 1sec. delay. Since, the 800-number is only accessible

I'm trying to write a simple function that does the following: [command] xify(5.2) [output] XXX.XX [command] xify(3) [output] XXX Any simple solutions (without using python/perl/unix script/...)? Thanks, Saghir --------------------------------------------------------- Legal Notice: This electronic mail and its attachments are i...{{dropped}}

I just changed the from DSL to Shaw Cable (static IP) configure the firewall but now asterisk I can not register with FWD nor VoipJet calls going out. I am using IAX with FWD Did I missed to change a setting? I don't think there is any though. I am on shaw extreme connection; I just talked shaw tech. and they are not blocking any port - I was told. So why IAX will not register with FWD and

I would like to implement enum lookup in my dial plan but searching for solution / implementation I'm getting confused what is current standard. On some pages I read that the ENUMLOOKUP is not in development anymore and suggesting on using Enumlookup.agi scrip , some are saying that Asterisk 1.2.0 comes with a new powerful ENUMLOOKUP. So there is probably no need to use this script anymore;

I've setup SPA-3000 and when the calls come through my phone is rining almost instantly but the [demo] doesn't answer till after about 13 seconds. So I have about 13 seconds delay and I don't know what setting is causing it; here is a part of my settings from extension.conf. [from_pstn] exten => 1000,1,Goto(demo,s,1) [demo] exten => s,1,Answer ; Answer the

Is anybody familiar with these IP phones AT-723 or AT-323 I think it is made by this company: http://www.atcom.com.cn/at723E.html -- #Joseph

After somebody records a message asterisk notifies me and encloses the WAV file. Though I'm not sure if this is a WAV format. I can not play it. According to the file specification it is: msg0000.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz How to play received message? -- #Joseph