similar to: deadlock

When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear F@510P) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power

My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem is : Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x861f6d8', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85a6420', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided

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

Hi, I'm not clear on how to use Power--over-Ethernet, specifically with Polycom phones. What I understand, is that by buying the Polycom 501 with the 802.3af cable bundle, I simply connect my phone, through the Polycom provided "special" RJ-45 cable, into a PoE capable switch, and voil?! Is this true? And if so, what happens when the Phone doesn't connect directly to the

In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN10000 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10

Hey everyone, I know this isn't a direct Asterisk issue, but some of you may know this answer. I recently upgraded the SIP version to 1.6.6 on all of our phones in the office. Everything is working fine, except one aspect. The phones in the office reboot randomly for no apparent reason. I haven't changed anything in the configuration files since the upgrade. The only setting in the

I ripped a rock-and-roll CD for a client's moh. But it's too loud. Is there a simple way to reduce the gain without having to remix the tracks? Thanks -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on "Teliax" before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi i have 20 people calling agents calling when ever they calling i get this below error May 30 00:46:57 WARNING[2840]: channel.c:785 channel_find_locked: Avoided deadlock for '0x8b2f50', 10 retries! and the voice go choppy, and voice breakages iam using Latest SVN, any suggestion to come over this problem ram -------------- next part -------------- An HTML attachment was scrubbed...

Dec 1 12:08:43 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! what does this

I would like to capture the lat/lon coordinates from a GPS-enabled cell phone or PDA. Is this possible? Must I subscribe to this information from the cellphone network provider, or can I capture it without charge? What devices will broadcast the coordinates? Is there a device that will broadcast its position inband that can be captured by Asterisk? Can an SMS message include coordinates?

Hi List, Not sure if this is the place for this so here goes ... We have a number of Polycom 501's connected to our * box and they work great. Some of our users have added a few entries into the directory on the phone. The problem is on those particular phones they now sometimes get "resource full" on the phone when accessing the directory. No central directory was configured.

I currently have Asterisk 1.2.7.1 and the Sangoma A200 w/ 6 FXO ports and HW Echo canceller. I have outgoing calling setup to use a group so that if one channel is busy it goes to one of the other channels. What's weird is that when I dial an outside number, sometimes it goes through and other times I get "You have reached an invalid pager number MCLL327." I have no idea what that

I would like to connect a Legacy PBX (Avaya IP Office 406) to an Asterisk Server. The Avaya has 3 CO Ports available. I thought buying a TDM30B card with 3 FXS ports to connect the * to the Avaya CO Ports. Is this the right approach? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions.... Daniel Kocher

I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an AGI, I get the "avoided deadlock" message below. *CLI> == Spawn extension (CONTEXT3, 6080, 8) exited non-zero on 'SIP/3211-1-081c40a8' -- Executing NoOp("SIP/3211-1-081c40a8", "") in new stack -- Executing AGI("SIP/3211-1-081c40a8", "diallocal.agi") in new

Hope someone can help. I have a situation where asterisk is sending a SIP CANCEL message before the Dial() timeout has hit. It doesn't always do it. Normally, we send an INVITE to the ITSP. They respond with a 100 Trying, then a 180 Ringing, or 183 Session Progress. It seems to be at this point that Asterisk starts the dial timer. Normally, when no more replies have been received by the dial

I just purchased a Grandstream gxp-2000, budgetone102 and a HT-386. Browsing to each device by IP address, I can get logged in using admin and I can see the advanced settings, however, if I try to change the settings and clicking the Change button, it just brings me back to ask for the password again.. I can't get into the Status page or any of the Account1-4 pages either. It just keeps

Dear users, Couple of days ago I have updated my * to 1.2.4 with ZAP 1.2.3 Since the upgrade I am having these problems: Feb 7 16:21:18 WARNING[387] channel.c: Avoided deadlock for '0x91a8b20', 10 retries!$ Feb 7 16:23:16 WARNING[16176] channel.c: Avoided deadlock for '0x91a8b20', 10 retries!$ Feb 7 16:23:28 WARNING[16176] channel.c: Avoided deadlock for '0x91a8b20',

Hi, I'm using the Read command the read a DTMF tone. In this read command I play a voice-file. But now when I press one off they keys of my telephone the voice-file will stop playing a the program go the next priority. Is it possible to play the voice-file until the right DTMF tone is pressed? (say for instance the Zero). Kind regards Arjan Kroon Mobillion B.V.

I just got a TDM30B, and the documentation doesn't indicate what driver I should be loading for it. Does anyone know? Any tricks I should know about before rolling this thing out? Thanks. B.