similar to: Storing voicemail in mysql

Hello, I have some problems to use G722, when my client sent an invite request to asterisk using G722/16000 codec asterisk respond with G722/8000 codec. I dont know exactly if Asterisk supports G722/16000 codec?? If yes how can I activate It?? Thanks. Rachid. Below wireshak trace:

Here's something that should be easy for RUBY pro's. Here is a script: 1.times do r = $agi.exec('DIAL', SIP/voipuser&Zap/32&Zap/33&Zap/34&Zap/35) r = $agi.get_variable('DIALSTATUS') # $agi.set_variable(' WHOANSWERED

Hi. I'm new here so I hope this is a sensible question/sensible place for it. I have a PSTN to IAX phone number with voipuser.org that I'm using to test an IVR service. The only trouble is that after approximately 40 seconds of silence (e.g. after background playback of a menu prompt) the call gets cut off. Is this a common problem? I've already set the ResponseTimeout to a big

I've just ordered a Siemens Gigaset C450 IP cordless IP/DECT phone, given that it's supported by asterisk http://www.voipuser.org/review_41.html However, I see that a slightly better Gigaset S450 IP is available for only a slight price premium. Are there any user experiences with the S450 IP? -- Eugen* Leitl <a href="http://leitl.org">leitl</a> http://leitl.org

Hi list! Again Wengo has made changes to their servers that require modifications to * configs. Is there anyone that has the 'new' wengo working with asterisk that could post their configs? Also they switched codecs, now G720a is required to connect. I can only find an (open) G729 codec, is this the same as G729a? Thanks!

Hi, I am simulating the sending of fax using sendfax through voip to reach an Asteria server via ZAP/1 ( PSTN phone line ) which then route call to a fax machine at ZAP/2. It seems like I am not able to establish any handshake with the physical fax machine using the sendfax program. Does anyone know why that happens and how to fix it? The scenario will be deployed in remote location in the

JID Pri S Owner Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host

Dear all, I am still not able to store voicemail into mysql and I am hoping someone can help me out. Here is my voicemail.cof: [general] format = wav attach = yes dbuser=ast dbpass=sqlpass dbhost=localhost dbname=asterisk odbcstorage=asterisk odbctable=voicemessages [default] ; Syntax for new entries looks like this: ; MailboxNumber => password,name,e-mail,pager,options ; (usually, the

Hi list! I have a strange echo problem. Two days ago I setup * 1.0.6. at a friend of mine. Just an * server and for outbound calls wengo.fr was used to place calls via sip. He had a strange echo on the line I didn't experience on my setup. Today I upgraded my asterisk 1.0.5 to 1.0.6 and suddenly I have an echo too on sip calls thru wengo!! I already verified wengo was not the source of

Hi, I'm searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all I have been googling and wiki-ing and have found a number of potential solutions to my questions, but I don't want to have to play about for too long and risk messing up my * box now I've just got it working, if one of you kind folk could offer your 2 penneth, (being a Brit I'll have none of this cents business ;] ). I want to send an SMS message whenever I get a voicemail

Asterisk must have a reasonably large community here in France judging from the number of people who came out to meet Mark. Either that or we were ALL there :) Something I've been waiting for, a voIP carrier on the models we are used to (low monthly or pay as you go, web account) has just set up their first beta test for 1 euro for the first month, 6euros if you decide to keep it. The basic

Hello there! I need some help to configure two Asterix boxes to route calls using SIP. I followed the instructions present at this site: "http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html", but I couldn't get it working so far. The only difference, besides the names that I've used, is that I'm using realtime to retrieve

Hi, I've been using nufone DIDs for months with no problem. Now there are codec problems that prevent any kind of calls working. For example, May 5 13:04:12 WARNING[928]: channel.c:2115 ast_channel_make_compatible: No path to translate from IAX2/NuFone@NuFone/25(256) to SIP/wengo-out-968a(4) May 5 13:04:12 WARNING[928]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't

Hi all, I am using Digium PCI board to receive PSTN call through regular phone line. It is no problem for me to receive calls, but I am not able to obtain the Caller ID if the calls are from the phone line. exten => s,1,Answer() exten => s, n, Verbose(1|incoming number is ${CHANNEL} calling to ${EXTEN} routing to ${phonenum} ) exten => s,n, Verbose(1|callid is ${CALLID(num)}) exten

I have a really baffling problem. A couple of months ago I purchased a pair of GrandStream GSX-2000 phones for use with Asterisk. At first all was well. But recently I've noticed terrible sound quality problems. Basically the sound will "glitch" or stutter randomly from time to time. Now, what is interesting is that this happens even with the phone totally disconnected from any

Hello gentlemen, I am new here. I have a D-Link DPH-80S SIP phone (it's a non-US model), and I am trying to make it work with Asterisk. I tried versions 1.0.7 and yesterday's CVS and the behavior is the same. The phone registers with no problem, and can accept calls. But when I try to make outgoing call, there is a series of invite requests from the phone, to which asterisk responds

Hello sip show peers does not mark hosts as NAT even though sip.conf and sip_peers table has nat=yes. spitfire*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status voipuser.org/gdsm 216.127.66.119 N 255.255.255.255 5060 Unmonitored 5560/5560 192.168.4.5 D N A 255.255.255.255 5060

PSI System Admin-Message-ID: <112664837810704203@ns1.psinetworks.net> Hi list members, I'm sure this question has been posted before but I am still unable to find the answer. I have a TDM 400P line card and I would like to set it up to IGNORE the distinctive ring pattern that I have for a fax machine. Many thanks Brad

I'm wondering if anyone out there is running a community PBX for their local Asterisk User Groups or area Linux groups. I've been thinking of setting one up but am stuck as to what services to provide that people would actually find useful. I know that I could setup simple SIP->SIP to allow everyone to call each other, but that's not generally too fun. -- Kyle Sexton