similar to: Segmentation fault due to SIP registration NUMBER 2

Hello, Don't know if this is related but I just got a segmentation fault today while trying to register my new SNOM200 phone: *CLI> *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '<sip:mattf2@10.10.10.15>' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from

Use oh323. Download the openh323 and pwlib tarballs from openh323.org Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY! good luck Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 sean.langley@gdcanada.com > -----Original Message----- > From: Senad Jordanovic [mailto:senad@cwcom.net] > Sent: Friday, September 12,

Hi! I am new to Asterisk too, I got the similar problem and I would like to know how to get * to work behind NAT. When I have the SIP Debug turn on, I got the error 479 from FWD when * try to register with FWD, it looks like * is using the local IP (192.168.x.x) in the Contact field. I have put the nat=yes in the [FWD.Pulver.com] content, but it does not seems to make Asterisk aware the

Hi, I have enabled ztdummy in order to have * compile it. I can modprobe ztdummy with no problems. The sole reason, i need ztdummy is to heve musiconhold and meetme working. However when I start *, it says this and does not start. ---------------------------------------------------------------------------- ---------------------- == Parsing '/etc/asterisk/zapata.conf': Found

Hi all, Anyone has experience using * through 128 kbs (or bigger) satelite link? In particular I am interested to hear how many calls could be put through 128Kbs satelite link simultaneously? Ta SJ

Hi List; Where I can download Asterisk GUI and what I can have benifit from it? Regards Bilal ____________________________________________________________________________________ Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545469

Make sure you are using G.711a, G.711u or GSM codecs.. I have not been able to get iLBC to work and someone the other days couuld not get SPX working.. You will need to enable/disable the codecs in X-Lite.. If you also want to control the codecs that * uses then put the following in the general section of your sip.conf disallow=all allow=alaw allow=ulaw allow=gsm Hope that helps.. > Hi,

Hello Everyone, About a month ago, someone put a question to the list about which ADSI spec to purchase from Telcordia. I looked in the archives, and it appears that this question was never answered, so I'll put it to the list in a slightly different manner: Do I need to purchase the Telcordia specs in order to learn how to write my own ADSI scripts? If so, which one? I found the Black

Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just "hangs in there". ATA is behind NAT, registers to an * with public IP

Are there any nice GUIs out there for Asterisk Realtime? Google doesn't yield much. I spent a day trying to get VoiceOne to work without much success. Thanks, Mike Clark

In United Kingdom, we have time based dialling pricing from most of Telco's based on time the call is placed! It is called PEAK (08.00- 18.00 Mon-Fri), OFF PEAK(18.00-08.00 Mon-Fri) and WEEKEND (all other times! Could someone from any of other countries let me know if time based charging exists in your country? Also, what numbers (up to 999) are commonly used for emergency, police or other

Hi, Do the callers in USA dialling from USA Telco lines always have to prefix the CITY/AREA code with "1" in order To successfully make a call to other USA destinations? ---- I have not been to USA (yet) :) Ta SJ

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News -

Hello! We are located in the UK, and we are planning to replace our old pbx with an asterisk based pbx. For outgoing calls our present pbx is connected to three PSTN lines which all have the same number. Internally, the pbx caters for quite a few extensions, and each extension can make outbound phone calls. Our telecom provider (your communications) gives us monthly itemized bills that list

Hi, all This is certainly very good news! http://www.neowin.net/comments.php?id=21119&category=main

hi all anyone that've heard of any working IAX hardphones yet? roy

Dunno if anyone else has seen this yet: http://www.scmagazine.com/us/news/article/563800/vulnerabilities+put+asterisk+telephone+systems+risk/ -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

I have recently noticed that druid @ http://www.voiceroute.org has created an open source edition of their platform. I downloaded it today and installed it on a play system where I have about 20 ip phones ranging from cisco, polycom and aastra phones. I didn't even have to configure them as the system automatically did it for me. I have been using trixbox/freepbx combination for over that last

Hello! How do I handle calls when they reach a queue that has no members? Currently, the callers are thrown out, because of the autofallthrough. The message is app_queue.c:2094 queue_exec: Unable to join queue 'queue-name' == Auto fallthrough, channel 'Zap/3-1' status is 'UNKNOWN' It seems that Queue() won't continue at a specific priority - like n+101 - if