similar to: RTP Binding Address

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

Does anyone know how to change the source IP address/Source Interface of RTP packets? Changing the SIP source IP address in sip.conf has no apparent impact on RTP. RTP traffic still uses the address assigned to the outbound interface.

I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for

I'm currently running asterisk to provide VoIP services to clients of the ISP I work for. I would like to be able to tell if I am loosing packets and/or are having other issues with any of the voice streams, so I can address them proactively. I'm not particularly interested in spending oodles of money buying one of the commercial analysis tools. Is there some open source tool (or

The question: Is it possible to change the RTP binding address? If no, does anyone have any ideas how to work around the problem? The network: 192.168.11.1 | < Asterisk > <Freeswan> < Iptables > | \

I've just had the most appalling performance from * ever. Dialling: Cisco 7960 => asterisk => IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 => asterisk => IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be

Hello, Cedric Hans has released an UNISTIM channel driver for asterisk (stable). You can download it at : http://mlkj.net/asterisk/chan_unistim-0.9.2.tar.bz2 Copy of README : This is a channel driver for Unistim protocol. You can use at least Nortel i2004 phones with it. Only few features are supported : Send/Receive CallerID, Redial, SoftKeys, SendText(), Music On Hold, Message Waiting

hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711

http://www.asterisk.org/node/48317 does a nice job of explaining the 1.4 jitter buffer, however it raised a question in my mind. In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels, are the UDP RTP packets renumbered on transmit, or is the original sequence number preserved in the UDP header? A comment is made on the referenced blog that jitter buffering is best implemented at the

Did Linksys really use Asterisk for the SPA9000 software? -- Andres Technical Support http://www.telesip.net

My phone is located behind a NAT, 172.16.0.0/21. Asterisk 16 is on a public IP. PJSIP has the config below: force_rport=yes direct_media=yes disable_direct_media_on_nat = yes direct_media_method=invite But when I send a call I see the RTP being sent to my private address, vs the public IP. This only happens when Asterisk has dialed the call to another carrier. If instead of Dial I choose

Hello, We have a system that interoperates with an external service, so that the basic call flow is: PSTN origination -> Asterisk A -> External service -> Asterisk B Initially the SDP from the external service tells the two Asterisks to send RTP directly to each other. Part way through the call the external service sends re-INVITEs both Asterisks to change the address for audio to

If there are multiple NICs in the box, how do we specify the local IP address to be used for RTP? Anything in rtp.conf ? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030609/d151f190/attachment.htm

Hi Joshua, Thank you for that. From the code it kind of looks like STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum: if (!ast_sockaddr_isnull(&rtp->strict_rtp_address) && STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(), rtp->rtp_source_learn.start)) { ast_verb(4, "%p -- Strict RTP learning complete - Locking on source address %s\n", Our call shows: #

I don't know in that specific output what happened. Your best course of action is to add further logging or step through the logic with all of the knowledge you have of the RTP streams to understand what is happening. On Mon, Apr 17, 2023 at 8:52 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Thank you for that. From the code it kind of looks like

DSL works by using the frequencies above 4k that were unused in POTS loops of yesterday. Load Coils, Bridge Taps, and DC taps are all devices added to lines to increase their reach and stability, unfortunately, they are DEADLY for DSL. Other problems can effect DSL service, and cause it to be 'flaky'. 1 Temperature, in Florida the large black cables are constantly beaten down by the sun,

Hi, Sorry for my bad english but I?ll try to explain my problem I got an Asterisk running in my house with ADSL... I?m using X100P and TDM400P cards.... My intention is get calls via PSTN to my house and Redirect to my computer in my work using X-Lite by SIP... Here?s the map with Firewalls Call for anyone to my house => PSTN => X100P => EXTENSIONS => SIP/RTP => ISA MICROSOFT

On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hello, >> >> Does anyone know if one of the "strictrtp" options disables RTP learning? >> As far as I can tell from the documentation the values "no" and

It's probably best if you read the logic[1]. There's an entire comment that talks about how it works. [1] https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 On Mon, Apr 17, 2023 at 7:10 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Could you confirm if the 5 second period for learning a new audio stream > is a minimum

I had a friend yesterday showing me his new T-mobile blackberry with WiFi Voip. I could not believe it until I actually saw him making calls. There is no T-Mobile cell coverage at my house but he was able to simply access the WiFi router and make the call. It appears this VoIP offering is tightly integrated since you use the same phone number to make and receive calls over WiFi or Cell.