Worm78

I'm currently using the Fax Pass Through mode, My SIP trunk provider (Teliax) does not support T38. I was on an older version of the CS410 and using the spa2001 and I had no issues. Unfortunately over the last six months I was battling a 4 second intermittent silence issue. That was recently resolved. It was a cable modem with ARP storm detection enables causing the issue. I still need to update my other 400 page post still. In that time, I upgraded to the latest software on the cs410, we changed the fax machine as the other one died, and I'm sure I have changed some settings on the PBX. All was well on the PBX side until we tried to fax. I worked with the SPA2001 for a week and reads horrible things. I upgraded to the ht286 and still no luck. Audio is crystal clear if I use the handset or a regular phoneo n the fax line. It is forced on g711u. We don't do much faxing and are ok with the occasional issue but I have tried 30 pages and not one has went through. Always a comm error. I hear the dee dee dee da shhhh and then i see the call online from the pbx for 60 more seconds or so. My guess it is some echo or silence supression issue. You mentioned this on another post but not sufre how to change on the cs410 "One common problem with the Gradstream devices is that they advertize UPDATE, but when they receive a UPDATE message with a SDP offer, they don't answer this. What you can do is to turn off the support for UPDATE. The setting is called "support_update" and is in the http://wiki.pbxnsip.com/index.php/Global_Configuration_File. I would try to set it to "false". ". Here is what Teliax says. I'm awaiting a reply from the support line. Is there anywhere in the CS410 to limit audio correction or clipping? They have an inbound fax service we will use for all inbound items. 5. Can I use my fax machine? Yes. In most cases you can use a fax machine but this service is not 100% guaranteed. VoIP uses compression technology to preserve bandwidth and send a high quality voice signal accross the internet. Part of compressing a voice signal involves removing the higher and lower portions of the voice frequency being sent over the internet. These frequencies are inaudible to the human ear but are very important to machines like faxes. We have some special instructions for faxing over the TelIAX network. Please email us at support@teliax.com for more information. Thanks, Brian

I had fax machine working for a while on an old SPA2001. I was troubleshooting another issue and now it does not work. I then upgraded to a Handytone 286 as the SPA was a bit iffy. ATA connects and works like a phone with no issues. I have the ATA and the pbx ext set to use G711. I have it set on pass through mode. Here is the log during the issue. It seems to try and handshake then goes silent. Any ideas? [7] 2009/04/10 10:50:29: Set packet length to 20 [6] 2009/04/10 10:50:29: Sending RTP for 6034554a414b8a2b@192.168.10.203#20f0a6c98b to 192.168.10.203:5004 [7] 2009/04/10 10:50:29: Set packet length to 20 [7] 2009/04/10 10:50:29: Receiving DTMF on codec 101 [7] 2009/04/10 10:50:29: Set packet length to 20 [6] 2009/04/10 10:50:29: Sending RTP for 1b1dee70@pbx#1336573554 to 63.211.239.14:30626 [7] 2009/04/10 10:50:30: Receiving DTMF on codec 101 [7] 2009/04/10 10:50:30: 1b1dee70@pbx#1336573554: RTP pass-through mode [7] 2009/04/10 10:50:30: 6034554a414b8a2b@192.168.10.203#20f0a6c98b: RTP pass-through mode [7] 2009/04/10 10:50:30: Call 1b1dee70@pbx#1336573554: Clear last request [7] 2009/04/10 10:50:32: Call 1b1dee70@pbx#1336573554: Clear last INVITE [7] 2009/04/10 10:50:32: Set packet length to 20 [7] 2009/04/10 10:50:32: Determine pass-through mode after receiving response [7] 2009/04/10 10:51:29: Set packet length to 20 [7] 2009/04/10 10:51:39: 1b1dee70@pbx#1336573554: Media-aware pass-through mode [7] 2009/04/10 10:51:39: Other Ports: 1 [7] 2009/04/10 10:51:39: Call Port: 1b1dee70@pbx#1336573554 [7] 2009/04/10 10:51:39: Call 1b1dee70@pbx#1336573554: Clear last request [5] 2009/04/10 10:51:39: BYE Response: Terminate 1b1dee70@pbx The fax is set to 14,400 , ECM is turned off. Error kicked back on fax report is comm error. Using the standard phone seems very clear and a good bi-way audio signal.

What was the fix for the HT-286 , I have a spa2002 I'm replacing with one due to intermittent faxing issues. Thanks

.200 is the pbx and .63 was my pc which x lite was running on. Is there something I need to do with wire shark differently?

I caught the issue happening while using wireshark. I started a call, opened Wireshark and went about my business for the day. 14 minutes in I was attempting to go to the palladion.net website to download the QOS tools suggested. The silence issue is visible in the capture about 80% of the way down. It appears is has soemthing to do with traffic routing with the website I hit. I'm no expert. Can I get a wireshark expert to take a look? Here is a link to the file. It was too big to attach. LINK I also noticed the netgear router had SIP ALG enabled. I disbaled this just in case it is messing with the softphones on the lan side.

I found out it is local as well. Not local as in house on the lan, but in the same area code. I ran an MTR trace on each trunk available. I found the one I was using was getting packetloss of 2-3% on a 15 minute scan. I switched to a new trunk from the same ITSP. I also forced the GSM codec instead of using 711. Internal and on trunk. Any other ideas? I plan on running the QOS items tomorrow afternoon.

So far it seems to be on long distance only. Any ideas?

It seems to be happening randomly all the time. I have the user now tracking out going / incoming, long distance / local and so on. I also just setup the email notification on the new version. The time it happens during the call is also random. It is usally in the first or 3rd minute.

That fixed the dial out issue. I removed the area and country code and outboundstarted working. We are now on the latest version. I made 5 calls of 10-15 minutes to my cell phone and placed the cell phone in front of a speaker for constant noise, had no cut outs. I also did this from multiple computers running xlite. Six calls later the following day the user had a cut out while calling a long distance number. It happened 2 minutes into the call and lasted 10 seconds. You can hear a dead silence come on when on the call. The local user just delays a bit so the customer on the other end doesn't realize he can't be heard. Any other ideas?

I did the update to 3143. Afterwards the incoming calls worked but outgoing calls did not. Here is what I received in the log when trying to call out. INVITE Response 404 Not Found: Terminate 887c3383@pbx It would change the number in front of @pbx of course. I checked all the dial plans and such. Trunks looked up and working. I didn't work on it all too much as I had some incoming DID lines not working as well. I spent a lot of time on the incoming issue first and come to find out a switch was down due to a patch cable being used by someone there on site. He needed a cable and borrowed one and it was the same time I was troubleshooting for 45 minutes. My method was I dumped the configuration files. Upgraded the PBX and restored the config files. I did noticed I now had duplicate trunks listed under dial plans when selecting a trunk in the drop down. Only one trunk was visible under trunk settings. Well besides the default pstn trunk. Should I have not restored settings? I also noticed it handles the alias number a bit different. It adds the number to the back of the account. I only have two DID lines and they both worked after I solved the missing patch cable mystery. Side note. On the lan side i found the router was port forwarding all ports from 20,000-60,000 as RDP. I changed this to port 3389 for now. This is lan side only and the 410 has no lan gateway. The pbx has its own static IP. Just wondering if this could have caused the one way audio issue. I also updated that router wgr614 v9 to the latest firmware while onsite. I will attempt another upgrade after I hear back and can schedule more downtime. Everything is restored to original and working for now.

I have a CS 410 that has ocassional one way audio issues. Loss of incoming audio. It picks back up 3-6 seconds later. No other quality issues. It happens on Snom 300 and Xlite phones. The cs410 device is currently public to the internet on a business dsl connection. Provider is Teliax. They suggest the following four codecs. G.711u (µlaw) - 80Kbps G.729a - 25Kbps GSM - 30Kbps G.726 - 50Kbps The PBX which is running software 3.0.0.2992 (Linux) I have a section under the phones and under the trunk to set a codec preference. Question is how do I put the codec inthe field? G.711U or G711U and so on.... My thought is try a few different codecs between the phones and pbx, and the pbx and the sip provider. Kinda out of ideas at this point. I looked at the link below but this version has a add / remove column. http://87.230.9.185/index.php?title=Trunk_...p;printable=yes The other issue is if I do try either version of entering the data as above I get a busy signal when I call in. My guess is I had it in incorrectly. Anyone using Teliax and/or Xlite phones suggest a certain codec? Should I go ahead and setup a codec for the local side to each extension as well? I visited this issue a while back at this post. http://forum.pbxnsip.com/index.php?showtopic=1194&st=20

Just got it There was a 9 in the prefix area...oops

No luck I did just realize something, took me a second and shouldn't admit it. The EXT i'm testing from is ext 121 Hence the 9 121 being sent to my cell. Is there a global setting for caller id anywehre? My old provider had it set and over wrote on their end so i never had the issue. Brian

I have the phone number in there. I still get 912-1 for the caller ID I tired with multiple settings such as

All is good with my Trunk setup. Thanks for the help. One issue I'm still having is caller ID. I only have one DID coming in on one trunk. One domain. There is an alias on it with my outside static IP. Broadvox said my From User is blank. Said it shouyld be in a sip.cfg on an asterisk system. Anyone know what that equates to in PBXNSIP gui? I have tried RFC settings with no luck. How did you get yours to work?

Thanks for the help but i can't view the picture. It is weird. It says I'm not logged in however It shows me logged in just above the error. Same issue from two machines after relogging in. Must be a forum bug. BR

The domain is named my public IP or the IP I connect to at Broadvox? Thanks, BR

I'm currently receiving (408 Request Timeout (Registration failed, retry after 60 seconds)) in the trunk list. I spoke to Broadvox and they said to leave the registration info blank. I have triedb oth ways but get the same results. I also have not changed the localhost to anything. Will changing this cause any other issues? I'm thinking I have the same issue as this guy. http://forum.pbxnsip.com/index.php?showtop...amp;hl=broadvox However he is using dynamic as he has a user and password. Thanks, Brian

I currently have a Teliax trunk setup and working using the registration type. I'm looking to switch to Broadvox due to constant outages and poor support with Teliax. I have setup an account with Broadvox. They have two types at which you can setup. Static which acts as a Gateway and needs registration turned off. Then dynamic which seems to work like the regular registering trunk. I know there is a few issues that were discussed in this post with the TO field. http://forum.pbxnsip.com/index.php?showtop...amp;hl=broadvox I have also seen a few posts regarding the domain name being local host and not the public ip causing issues. Question is, is there a way to disable registraton on that specific trunk and not across the board, or can you even disable it across the board. This would work long term but not during testing as this would eliminate the ability for me to keep my Teliax trunk up. I have looked at the tips on the wiki for broadvox but it doesn't seem to help. Currently running version 1.5.2.10a Thank You, Brian

Anyone else seeing trunk issues using Teliax? They have had some ups and downs for the last month. I move to another Trunk and everything is fine. That list of OK trunks started to dwindle. They explained they were just doing some updates. Today they told me they did the final upgrade on the last lax.teliax.net trunk I was using. Since then I cannot connect. They have said several Asterisks users who had to make some configuration changes. I currently get a 481 Call Does Not Exist (Registration failed, retry after 60 seconds). I have the information correct. The selected server is correct on the dashboard provided by Teliax. Anyone else figured out what to change? They can give me nothing more then invalid user ID. Any ideas? note the call id: field. Teliax thinks this is the issue. Below is the log Logfile Clear or Reload the log. [2] 20081119145322: Web interface triggered reregistration of trunk 9 [8] 20081119145322: route_pending_packet -1097: entry=url sip:lax.teliax.net [8] 20081119145322: route_pending_packet -1097: entry=naptr lax.teliax.net [8] 20081119145322: route_pending_packet -1097: entry=srv tls _sips._tcp.lax.teliax.net [8] 20081119145322: DNS cache_lookup: dns_srv _sips._tcp.lax.teliax.net -> [8] 20081119145322: route_pending_packet -1097: entry=srv tcp _sip._tcp.lax.teliax.net [8] 20081119145322: DNS cache_lookup: dns_srv _sip._tcp.lax.teliax.net -> [8] 20081119145322: route_pending_packet -1097: entry=srv udp _sip._udp.lax.teliax.net [8] 20081119145322: DNS cache_lookup: dns_srv _sip._udp.lax.teliax.net -> [8] 20081119145322: route_pending_packet -1097: entry=a udp lax.teliax.net 5060 [8] 20081119145322: DNS cache_lookup: dns_a lax.teliax.net -> 74.201.8.23 [8] 20081119145322: route_pending_packet -1097: entry=udp 74.201.8.23 5060 [8] 20081119145322: Trunk 9 (Teliax 1) has outbound proxy udp:74.201.8.23:5060 [8] 20081119145322: route_pending_packet -1098: entry=udp 74.201.8.23 5060 [8] 20081119145322: Send Packet REGISTER [9] 20081119145322: SIP Tx udp:74.201.8.23:5060: REGISTER sip:lax.teliax.net SIP/2.0 Via: SIP/2.0/UDP 75.149.200.210:5060;branch=z9hG4bK-1a001a5300f78f19b8522ea17d553fa4;rport From: "brichter" <sip:brichter@lax.teliax.net>;tag=26524 To: "brichter" <sip:brichter@lax.teliax.net> Call-ID: ttagjj3g@pbx CSeq: 124 REGISTER Max-Forwards: 70 Contact: <sip:brichter@75.149.200.210:5060;transport=udp;line=45c48cce>;reg-id=1;+sip.instance="<urn:uuid:159f4fe2-2ba5-48e2-af0c-549b66b46747>" User-Agent: pbxnsip-PBX/1.5.2.10a Expires: 3600 Content-Length: 0 [9] 20081119145322: SIP Rx udp:74.201.8.23:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 75.149.200.210:5060;branch=z9hG4bK-1a001a5300f78f19b8522ea17d553fa4;rport=5060 From: "brichter" <sip:brichter@lax.teliax.net>;tag=26524 To: "brichter" <sip:brichter@lax.teliax.net>;tag=vyegX37aDjQHc Call-ID: ttagjj3g@pbx CSeq: 124 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="lax.teliax.net", nonce="2e0a5f73-73b6-dd11-8406-00114336c25a", algorithm=MD5, qop="auth" Content-Length: 0 [8] 20081119145322: route_pending_packet -1099: entry=udp 74.201.8.23 5060 [8] 20081119145322: Send Packet REGISTER [9] 20081119145322: SIP Tx udp:74.201.8.23:5060: REGISTER sip:lax.teliax.net SIP/2.0 Via: SIP/2.0/UDP 75.149.200.210:5060;branch=z9hG4bK-b18664ba5bb745a6d3828102d36620bb;rport From: "brichter" <sip:brichter@lax.teliax.net>;tag=26524 To: "brichter" <sip:brichter@lax.teliax.net>;tag=vyegX37aDjQHc Call-ID: ttagjj3g@pbx CSeq: 125 REGISTER Max-Forwards: 70 Contact: <sip:brichter@75.149.200.210:5060;transport=udp;line=45c48cce>;reg-id=1;+sip.instance="<urn:uuid:159f4fe2-2ba5-48e2-af0c-549b66b46747>" User-Agent: pbxnsip-PBX/1.5.2.10a Authorization: Digest realm="lax.teliax.net",nonce="2e0a5f73-73b6-dd11-8406-00114336c25a",response="601290b0c34cfa273a4ca9dfdf94ce45",username="brichter",uri="sip:lax.teliax.net",qop=auth,nc=00000001,cnonce="edb10a67",stale=true,algorithm=MD5 Expires: 3600 Content-Length: 0 [9] 20081119145322: SIP Rx udp:74.201.8.23:5060: SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 75.149.200.210:5060;branch=z9hG4bK-b18664ba5bb745a6d3828102d36620bb;rport=5060 From: "brichter" <sip:brichter@lax.teliax.net>;tag=26524 To: "brichter" <sip:brichter@lax.teliax.net>;tag=vyegX37aDjQHc Call-ID: ttagjj3g@pbx CSeq: 125 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Content-Length: 0 [5] 20081119145322: Registration on trunk 9 (Teliax 1) failed. Retry in 60 secon

Quick Update After turning of UPNP on the router all has been well. It seemed the intermittent problem was when upnp did a scan every 30 minutes. I changed the scan time to a shorter interval and the problem increased. Thanks

Seemed to be working fine from wireless and from the Snom phone. Should have another update by them iddle of the week. Thanks

Other quick note so I just switched back to lan and it happened 2 out of 4 calls. SO it seems wireless is not having the issue. It is a netgear 614 router. Any ideas.

Update: Issue is still happening but not as much. I connected via wireless and tried and couldnt get it to happen in a time frame in which it normally happens 5 times so I decided to run a wireshark. When on wireless I don't get checksum errors on my wireshark dump. If on the lan I do. I get these same checksum errors on my personal working system as well. I have attached both. I also removed the switch from the picture to try to eliminate the check sums. They were still there. I then removed the router and only used the switch and the check sums remained in the picture. other changes I have X lite pointed directly to the pbx for DNS to help ensure the router is not being used. I also setup a snom desk phone and it seems to be working well but the user hasn't used it heavy. Hopefully a full week of using the phone will show if the issue is gone or not or only while using xlite. I have also used the dump feature on X lite but can't get the problem to happen during that of course. Any ideas from looking at the calls made during the attached wireshark dumps?

I'm still having issues if anyone has any ideas. I moved the lan port of the PBX from the router which is attached to the switch, to directly to the switch. All four router ports are open on the lan side as it is just a home type router. It seems to help the checksum errors a bit but I'm still getting them during silence. I used to always get the checksum errors and even more so during silence, however now I mainly only get them during it connecting and then it comes in with long silence periods. I did change xlite to conserver bandwidth during silence periods and this helped a bit too. Still having the issue though. Nothing so far has impacted the original issue in duration or quanity. I'm attaching a cheesey drawing to help with the layout. Any other ideas? GW on the lan side pbx connection settings is removed. I run a Workstation version of pbxnsip on my personal network and I see the similar checksum errors in wireshark. Thanks SKMBT_C45108090312590.pdf