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...e. Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter x.x.x.x 5531341d06b 00:07:42 0000023123 0000063836 (73.41%) 0.0000 0000023102 0000000000 ( 0.00%) 0.0007 Peer IP changed to protect the innocent :-) ________________________________ From: tjrlist at live.com<mailto:tjrlist at live.com> To: asterisk-users at lists.digium.com<mailto:asterisk-users at lists.digium.com> Date: Mon, 19 Jan 2015 12:17:25 -0600 Subject: [asterisk-users] sip show channelstats reliable? I am seeing lots of lost packets when running the command sip show ch...

I would recommend capturing traffic outside your Asterisk server with Wireshark, then running the Telephony/Rtp/Analysize Streams option to determine if you have packet loss at that point in the network. On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <tjrlist at live.com> wrote: > Thanks but no Adtran here. > > I do think these stats are indicating an issue, I just don't know how to > prove it outside Asterisk. > > > ------------------------------ > From: EWieling at nyigc.com > To: tjrlist at live.com; asterisk-users...

...OFFTOPIC openvox (ricky gutierrez) > 7. Re: SEMI-OFFTOPIC openvox (A J Stiles) > 8. Re: MWI issue (Haley,Scott A) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 19 Jan 2015 12:17:25 -0600 > From: Todd R. <tjrlist at live.com> > To: Asterisk-Users List <asterisk-users at lists.digium.com> > Subject: [asterisk-users] sip show channelstats reliable? > Message-ID: <BLU173-W265CCDC9CB89501E36210ECD4A0 at phx.gbl> > Content-Type: text/plain; charset="iso-8859-1" > > I a...

...t digium.com > wrote: > I would recommend capturing traffic outside your Asterisk server with > Wireshark, then running the Telephony/Rtp/Analysize Streams option to > determine if you have packet loss at that point in the network. > > On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <tjrlist at live.com> wrote: > >> Thanks but no Adtran here. >> >> I do think these stats are indicating an issue, I just don't know how to >> prove it outside Asterisk. >> >> >> ------------------------------ >> From: EWieling at nyigc.com >>...

Trying to figure out the best way to pull an active call out of a queue by unique id and put it on hold. I don't want to put it on hold on the agent's phone but I want it to be pulled away from the agent's phone and into Asterisk limbo somewhere. Shortly after I want to pull the same call out of limbo and redirect it back to either the same agent or another. I was thinking about call

I am seeing lots of lost packets when running the command sip show channelstats at the CLI. There are issues across multiple Asterisk servers I am trying to diagnose but everything I read seems to point to this command being pretty unreliable. Can I trust the info this command shows? I am showing lots of lost packets in sip show channelstats but I can't see any packet loss when pinging the

Thanks but no Adtran here. I do think these stats are indicating an issue, I just don't know how to prove it outside Asterisk. From: EWieling at nyigc.com To: tjrlist at live.com; asterisk-users at lists.digium.com Date: Mon, 19 Jan 2015 13:55:33 -0500 Subject: RE: [asterisk-users] sip show channelstats reliable? I?ve seen something similar with Adtran SIP gateways. When a re-invite happens the Adtran gets all confused about call stats and marks the pre-rein...

Right now we I am using Asterisk boxes as a gateway between our Level 3 SIP trunks and our customer PBXs. I love and understand Asterisk but the company I am working for is looking for a more "Commercial" type solution where we can go to a vendor for support etc. I know, we can get Asterisk support etc.. It's not my decision and I sort of get why they are leaning away from Asterisk,

Does anyone know how to set the music on hold class with the Manager Interface in 1.8? Here is what I am using but I end up just getting no music when I put this in place, when I remove it the default is back. The classes I am setting work elsewhere just fine. I did not include the opening of the socket, logging in etc because that's all working fine along with other things I am doing within

...Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter x.x.x.x 5531341d06b 00:07:42 0000023123 0000063836 (73.41%) 0.0000 0000023102 0000000000 ( 0.00%) 0.0007 Peer IP changed to protect the innocent :-) From: tjrlist at live.com To: asterisk-users at lists.digium.com Date: Mon, 19 Jan 2015 12:17:25 -0600 Subject: [asterisk-users] sip show channelstats reliable? I am seeing lots of lost packets when running the command sip show channelstats at the CLI. There are issues across multiple Asterisk servers I am t...

Took me a while but I have finally embraced cloud computing and all the benefits. The only thing I have yet to feel comfortable about putting in the cloud is real live Asterisk boxes to be used in production. I know it's being done because as far as I know Twilio is using Amazon for their Asterisk boxes. I have read all the fun articles on building hobby type systems and that's all great.

I currently have all agents/members logged in with local channels. When a call is sent to one of the agents, then the agent transfers the call out the line frees up on their phone but still shows in-use until the call that was transferred is hung up. How can I free up the agent/local channel when the call is transferred? This is a huge problem because the agent can no longer receive calls on their

Been working with Asterisk for a long time but this is the first time I have dealt with this issue. I am setting up an Asterisk box (FreePBX not my choice) to interface with an e911 provider. They say their switches only listen for RTP on ports 20000-21001 which is outside the normal range Asterisk listens on 10000-20000. I wish I knew more about this topic but since I have never had an issue

OK, been messing with Asterisk for a long time and I have my opinion on where the issues lies but sometimes it's just nice to see what others think that can relate :-) Here goes.. Inbound calls flow like this:Tier 1 Provider (SIP) > Asterisk 1.8 > Name Brand PBX - Calls work fine Outbound calls flow like this:Name Brand PBX > Asterisk 1.8 > Tier 1 provider (SIP) - Calls work fine

To be sure you could setup a soft phone and see if the caller ID name comes in correctly. > On Mar 10, 2015, at 8:41 AM, Jordan Cook - Gyron Networks <jordan.cook at gyron.net> wrote: > > Hi, > > In my dialplan I have the following line. > > same => n,Set(CALLERID(name)=Support) > > I am expecting this to always set the caller id name to ?Support? -

Just put a new phone in place with the latest firmware from Cisco. This is the first SPA501G we have with this firmware. In the Asterisk CLI we are now seeing the error message below about once every second. When we unplug the phone, the messages quit. NOTICE[15539]: chan_sip.c:9602 copy_header: No field 'CSeq' present to copy Thanks in advance for any assistance on this.