search for: g711u

...GE----- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone INVITE (G711U) cseq 2 From NexTone 100 Trying cseq 2 From Asterisk 200 OK cseq 2 From Asterisk ACK cseq 2 From NexTone 200 OK (711U) cseq 1 From Asterisk ACK cseq 1 From NexTone Call continues until one side hangs up... Failed Call: Call...

...he 'Flow' diagram from Wireshark from a tcpdump on the OpenSIPS host follows, hopefully the email clients will not mung it too much. |Time | Client | Asterisk | | | | OpenSIPS | |7.158764 | INVITE SDP (g711U g7 | |SIP From: "760xxxxxxx" <sip:760xxxxxxx at client To:<sip:317xxxxxxx at OpenSIPS | |(5060) ------------------> (5060) | | |7.159003 | | INVITE SDP (g711U g7 |SIP Request | |...

...4XXX they can dial *98 and PSTN calls, and yes they are all in the same context since April 2006! SIP to PSTN - OK SIP to IAX - OK This is a graph from ethereal: Dialing 4214, my own SIP extension! |Time | 192.168.34.26 | XXX.XXX.XX.XX | |11,219 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:4214@194.117.36.75:5060 To:sip:4214@XXX.XXX.XX.XX:5060 | |(2752) ------------------> (5060) | |11,721 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:4214@194.117.36.75:5060 To:s...

...ing phone-->asterisk-->PRI. I've not changed the configuration files during the upgrade. sip.conf is: allow=ulaw allow=ilbc allow=g726 allow=g729 allow=g723.1 And all the phones had been using ulaw before. The audio is not bad.. but sounds much more like a cell phone (Even with ulaw G711u) and 'under water'. Any thoughts?

Hello all, I've got an AVM Fritz!Card PCI that I'm using with Asterisk under chan_capi (0.3.5) Phones on our internal network all use g711u. I'm aware the chan_capi uses g711a by default. To reduce the need for transcoding, I decided to make everything use the same codec. First I changed all the phones (Cisco models 7905G, 7940G and ATA 186 all running SIP) to g711a, but it seemed to break the echo suppressor in the ATA 186, so t...

...the SIP call to another SIP gateway using Dial(). Even though codec preferences have g729 listed first, it never gets used. Both gateways have separate peer entries in sip.conf, and both have canreinvite=yes set. Can Asterisk change the media type during a re-Invite? The call is answered as g711u initially, and then Asterisk plays a menu, and then does a Dial(). I can see Asterisk doing the reInvite, but the protocol stays at g711u. Tom

Hi, How do I specify a codec to use for a SIP call? IE.. If I'm doing Dial(SIP/blah) for some reason the call is connecting using the codec at the bottom of my allow list rather then top (G711u)... and I'd like to force it to G711u if possible.

I installed G729 from Diguim and I was expecting the sound quality on my i686 machine to be better than gsm. Compared to gsm, G729 sounds closer and a little robotic. Is this what is supposed to be or am I missing something? I am interested in G729 because the internet in my country is very expensive and I want to save every bit possible. I want to use G729 because it takes less bandwidth for

Hi, Does anyone know what I need to do to get these: http://www.enicomms.com/cutglassivr/ Sounds files to work? I've tried loading them, but they are completely silent (format mis-match maybe?). Specifically, when I try to enter voicemail, nothing plays... though it clearly tries. I'm looking for replacement sound files for the default Allison, as I feel she is kind of breathy. I have

...93.414 | 100 Trying| |SIP Status | |(5060) <------------------ (5060) | 7 |195.529 | 183 Session Progress SDP ( telephone-event) |SIP Status | |(5060) <------------------ (5060) | 7 |195.536 | RTP (g711U) |RTP Num packets:969 Duration:19.360s SSRC:0x74A81A70 | |(10024) <------------------ (12350) | 7 |195.689 | RTP (g711U) |RTP Num packets:959 Duration:19.220s SSRC:0x5614CA99 | |(10024) ------------------> (12350...

...situation comes and goes throughout the call. Bandwidth isn't an issue as I have a 3MB/1MB connection and there is at most 2 concurrent connections. Also using pingplotter to monitor iax2.sixtel.net shows little or no packetloss. Just as further info, I am using a SPA-2000 to connect to * with G711u as the preferred codec. Anyone else experience the like or have any suggestions on what may be causing this or ideas on how to debug? Brian

I've got the XTEN Lite soft phone mostly working with * but it's dropping out like a very bad cell phone call. The GSM codec is worst (unusable), G711u and G711a are best but not good enough to use. I don't think it's a lack of bandwidth. What tuning options or approaches should I be investigating to make this work. Also, what's the best soft phone(s) for Windows XP? thanks, -reed

Do somebody using CISCO AS5350 with Asterisk? Which protocol do you using: H323, MGCP, SIP? This direction: [12sp->Asterisk->h323->as5350->isdnPSTN] is ok But reverse: [isdnPSTN->as5350->h323->Asterisk->12sp] cannot hear 12sp, but 12sp hear PSTN (codec g711u) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041004/759555de/attachment.htm

Anyone have an example XML file for the PAP2T? Cheers, j

Ok I have an * server finally setup and acepting calls from freshtel and I am VERY impressed at how well the Freshtel.net service works but thats another subject :) I have it all setup so that I can Dial my DID number on freshtel and that gets set to my * via IAX. At the moment I have the demo configured so that I can test it all and make sure it is all working. The problem is that I

...oanswer": What is required from the user agent, such as a SIP phone, to be able to hear the playback without Answer()? I'm asking this because when I used X-Lite, I could hear the the audio file but when I used a hardware phone (an ATA in fact) I couldn't hear it. The ATA supports G711u, which is the codec I set for the peer. Only when I change the dial plan to Answer(), then Background(file, skip)in extension.conf could I hear the audio using the ATA. Seems if the channel must be answered for it to hear it. Is there something required from the ATA to support the noanswer feat...

...but still scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323 EP and it's ok. And from now, it's also ok when H323 EP call SIP one's! No need to say that H323<->H323 is working, as well as SIP<->SIP. Running CVS version from yesterday. Used codecs are G711U & A, G723.1 and G729. If I just use G711 it's the same. SIP EP has to call first when * is started to make it work. Any hint? Also, H323 is still broken and working without FastStart. Is there a workaround existing? Regards -- Daniel

...e to login successfully. The two phones plus the Asterisk system are all on the same LAN with private addresses assigned to each of them. When a call is initiated and is picked up on the other end, there is completely no sound at all (as in the line goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and SPX. >From the Asterisk CLI I see the following errors; i) Unknown RTP codec 72 received ii) RFC3389 support incomplete Anyone got ideas on how I can go about this? Thanks in advance. Julius Kidubuka "When you do the common...

...e two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use the default g711u (ulaw) codecs. Adjusting the jitter buffer and jitter level settings to various values did not help. We are running Asterisk 1.2.1 on Centos 4.2 (Linux 2.6x kernel) on a dual-processor Dell Poweredge 2850 server with 1 Gb RAM. This machine has a TE-210 Dual-T1 card plugged in. The meetme.conf file...

...- ; Specify and configure CODEC related ; options ;----------------------------------------- [codecs] ; ; Define the codec list of the channel driver. ; Every "codec" option may have a "frames" option ; associated with it. ; Valid values for the "codec" option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3 - G.723.1(6.3k) ; G72315K3 - G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726 - G.726(32k) ; G72616K - G.726(16k) ; G72624K...