similar to: Nominal Jitter buffer Configuration.

Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Problem: Make a call on a Polycom 320 IP phone to

I am debugging an intermittent issue of missing audio on calls that come from a SIP provider into our asterisk-11.10 installation. Sometimes, incoming calls from this provider work correctly, with audio streaming in both directions. Other times, with the same setup, the calling party is unable to hear the IVR recording from the asterisk installation, although in fact the streaming is supposed to

Hello Everyone, Today we observed asterisk sending two invites for the initial call before the call was established (ie, not re-invites). There were no changes made to the configuration for a very long time, and was kind of confused when seeing this action. Can someone please suggest where to look to remove this behaviour? U 2014/08/12 07:34:20.405029 192.168.2.10:5060 -> 192.168.2.20:5080

Hi all. I am having lots of trouble with random calls dropping after 20 seconds, and I finally managed to capture a full sip trace. I'll paste it in full below, but I'll give a summary first. It seems that Asterisk is not recognizing the ACK messages that it receives from the Grandstream ATA. This happens only on the ACK that follows the OK that marks a call as established. This makes

Hello, I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call. When a fax call is made Movistar send only T38 in the INVITE. Invite example: v=0 o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 s=sip call c=IN IP4 192.168.1.2 t=0 0 m=audio 6370 RTP/AVP 18 101 a=fmtp:18

Hello, I am trying to setup a Grandstream GXP2160 IP-phone with secure calling (SRTP). Secure signaling SSIP for registration is working great ! I follow this guide : https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial But when I try to make a call with SRTP, I get stuck. There is an initial INVITE which is anwered with a 401. There should follow a new INVITE with a nonce,

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

im having a strange problem related to call-limit for peers. well im not sure if its related to call-limmit or not. Bottom line is: I call a user A, from user B. user B hears silence, untill it goes to voicemail. when user B hangsup. user B's call limit is reset to 0 but user A's call limit is not reset.strange thing is user A's status on cli is shown as NOANSWER, while user B did not

Hi All, I have Asterisk 11 talking to Avaya over SIP trunk using TLS and SRTP. On incoming calls from Avaya asterisk complains of 'unsupported crypto parameters: UNENCRYPTED_SRTCP' and rejects the call with '488 Not acceptable here' Doesn't Asterisk support UNENCRYPTED_SRTCP as crypto parameters in sdp? FYI SDP looks like this. v=0 o=- 1429194215 1 IN IP4 XX.XX.XX.XX s=-

On 11.9, trying to set up a callcentric peer: sip debug: > <--- SIP read from UDP:204.11.192.161:5060 ---> > INVITE sip:1777<myccid>@10.10.11.180:5060 SIP/2.0 > v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46aaaaef4249814d16a2ffb990d > f: <sip:<calling number>@66.193.176.35>;tag=3606475083-968127 > t:

Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 100 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP

I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default

Hello friends: I am facing cutoffs randomly when negotiating calls. The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the remote peer receives a ghost call. Along the atempt I could see six times a messages regarding NAT isuues *[3]* I hope anyone can give me an

Hello: I have this situation: I can make calls internally, I can make inbound calls but I can't make outbound calls. Thanks in advance. These are my devices: * asterisk 11.8.1 = 192.168.1.22 * sipphone grandstream gxp2160 = 192.168.1.5 * gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4 port 1 (FXS) connected to an analog phone port 3 (FXO) connected to the PSTN These are my

Hi Guys, I have Spikko setup as provider of DID and outbound routes and I can make calls out but no inbound calls via DID can be made. I did a sip debug which is reported below. I never receive the call though, I have a catch all in my inbound routes and it doesn't hit my context at all or not sip invite comes in: FreePBX: Trunk Name: *Spikko* Peer Detail *username=MyUsername*

Le 09/09/2016 ? 18:32, Madushan Geethanga a ?crit : > Hi, If you're not using RTP encryption did you uncheck the option in your RTP TAB from identity ? > > This is the log. ex dialling 0 from snom phone > > > <--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878 > <http://123.231.72.210:33878> ---> > INVITE sip:0 at 54.206.59.252

Greetings all, First time poster, Sorry if this has been answered here before. We recently replaced a failed 1.4x asterisk PBX at a customer location. Voicemail access was setup when the customer dialed *8, This worked in 1.4. Now, Running 1.6 (I know it's old I had to load it quickly, And that's what I got working first. It'll get upgraded to 1.8 soon). The strange part is *8 no

I am having a problem with one of my callers who is using either g729 or alaw.  I can do alaw but not g729 so asterisk should negotiate alaw right?  In fact from the sip debug it looks like it does, but then I get the dreaded "channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the call hangs up.  Why? Last minute thought: Is it possible that

I've been running the 8/1/2004 Head release up until a little over a week ago. I was forced to due to a card failure to upgrade to 1.2.16 without any advance preparation or testing (most of my connections are via satellite to all corners of the globe with high latency). Up until the upgrade I was running with very few issues. Since the upgrade I have been experiencing strange issues

Hi I request your help because i don't have actually a solution at my problems. I have a Asterisk Server in 1.6 Connected at a SIP Provider This provider supply me 2 numbers: 003318364xxxx (official number) 081169xxxx (Nddi Number) When i receive a call on the 081169xxxx, he don't use the extension. He use the 003318364xxxx extension. SIP Debug: <--- SIP read from