similar to: Asterisk codecs and packet size

Hi, We observed the packet size used in asterisk is about 20 ms. We would like to know if is possible to change this value to 10 or 30 ms for example. If so, how could I change it? Thanks in advance and best regards __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Google has given me too many responses, so I'll ask the list: Where in the Asterisk rtp source code can I find the default millisecond frame size? I've looked around for obvious pointers, but it's not clear. I'd like to "force" my Asterisk server to use a certain frame size all the time. (Of course, ideally I'd like to prefer or even force that frame size in a

Hello! I'm facing completely choppy sound. The wireshark trace shows, that there are a lot of codec changes without any trigger (means no options or reinvite or any other package). Background: The call is initiated by asterisk and is received by the same asterisk conference room via Phone extension -> asterisk -> provider A -> provider B -> asterisk. Asterisk initially sends

Hi, there is some progress, even I would'nt call it that. At least they admitted it's caused through some changes from their side. @Rowland: Remember that "old Samba method" part? This is their answer. I don't know what to make of it. Maybe someone with more knowledge about the develoment of Samba can give me a hint:

Hi, try to setup asterisk as an ISDN2H323-Gateway. The only problem i have after establishing a call is, that Audio works only from IP to ISDN-Phone but not from ISDN to IP-Phone. A known problem ??? Thanks in advance Michael i am using asterisk-cvs, pwlib V1.6.6 (janus), openh323 V1.13.5 (janus) and oh323-0.6.0 Here are my config's ############## # modem.conf # ##############

Hello, what i want to do is to find a way how i can solve the following problem. we want to offer our customers in the country side also isdn over voip but we have to use internet connections from another company for this. This company offers a QoS on this connections but only with 192kbit bandwith and with the ATM headers a normal g711a call has exactly 103,5 kbit/s so we can only use 1 channel

Hi, I am running FC1 with latest patches of 3/30/04, and I have the latest CVS as of this morning 3/30/04 of asterisk, zap and libpri. The SIP device I am using is a Sipura SPA-2000 with G726-32 "Forced". When I 'make clean" and recompiled zaptel, libpri, asterisk and start asterisk I can see: [format_g726.so] [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data) ==

Hi, is there a way to use my Belkin F6H650 on the USB port? I already installed usbhid-support, but neither the hidups worked (reports "/dev/usb/hiddev0 is not a UPS") nor genericups with upstype=7 and the hiddev set as device ("ioctl TIOCMSET: Invalid argument") worked. I have installed the debian sarge package of nut 2.0.1 (2.0.1-4) Regards, Philipp

On 29/06/2023 07:38, Ingo Asche via samba wrote: > Hi, > > there is some progress, even I would'nt call it that. At least they > admitted it's caused through some changes from their side. > > @Rowland: Remember that "old Samba method" part? > > This is their answer. I don't know what to make of it. Maybe someone > with more knowledge about the

Hallo, just my 2 cents: So Samba 4.12 works, but 4.13+ doesnt? Maybe you can use the same strategy here as used for Win XP or older OS: Setup an isolated (virtualized?) DC with samba 4.12 just for the synology to connect to? You could use firewalld/ufw rules to only allow traffic to the samba ports from one single source IP-adress (the synology) to limit the exposure... Just until synology

Hello list, Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle. Grandstream allows for 8 different codec specifications. I have defined them as 4 x G726 & 4 x alaw. Snom allow for 7 different codec specifications. I have defined them as 3 x G726 & 4 x G729. The SIP peers are both defined as : disallow=all allow=g726 allow=alaw allow=g729 allow=gsm This is the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the

Hi all, I have an Asterisk server connected in a PABX (TELEDATA) by channel Unicall.. I`m having problem when somebody call from PABX to Asterisk.. Eg: When somebody dial 1234, I received 1111112222333333444444 in the Asterisk CLI... If somebody can help me... or already saw this... Everton Goularth Uberlandia - MG - Brazil _______________________________________________________

Yep I did the same. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Francesco Peeters (Asterisk) Sent: Saturday, 21 January 2006 5:34 PM To: fbraeuer@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

Hello, I'm wondering if anyone has benn able to successfully get g726-16 passthrouhg to work? I am wanting to use this codec instead of g729 as I'm running out of DSPs using a high complexity codec on the Ciscos. I would think it would work just as g729 does, which has been working fine for me, but it does not. G726-32 does work great however, but it's like Asterisk doesn't

hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? -- --------------------------------------- Marek Cervenka =======================================

Hi everybody, A customer requires G726-40 with Asterisk... I know G726-32 is pseudo-standard, but he definitely wants G726-40... Is there any (easy) way, to integrate G726-40 into Asterisk? Has anyone done this before? Any hints? Please help! Due to a misunderstanding, my product manager already offered this to the customer and now i do not know how to do it... Thanks a lot in advance,

With sipura (I tried this with both the 3000 and 841) set to prefer the g726-32 codec, a call from the sipura to asterisk will use g726. Asterisk sip.conf has: disallow=all allow=g726 allow=gsm allow=alaw When the call is from asterisk to the sipura, asterisk will not use g726. It ends up using alaw. I usually use stable but I tried this with head too, and same thing happens. Anybody know how

Hello, Is there a way to change the format from the default? 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format ${MEETME_RECORDINGFORMAT}). Default filename is meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. - requires chan_zap.so Many thanks ******************************************************************** This email and any attachments

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --