[asterisk-users] sangoma a102d + Asterisk 1.2.14 ... bridging together 2 call legs on same PRI?
Hi all, Are there any issues to be concerned about when calls come in from PSTN to a PRI card and are forwarded back out the same PRI card? Anything different have to be enabled in zaptel.conf or zapata.conf or the Sangoma configs to make this work? What about using .call files that join two ZAP channels? Channel: ZAP/1/4081234567 MaxRetries: 0 RetryTime: 60 WaitTime: 60 Application: Dial Data: ZAP/1/4083456789 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] AMAFlags alwaysDocumentation (or 3 inastcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill)
Hi Folks, I figured out how to get the AMAFlags set correctly! It was not documented anywhere I could find, and I grepped through the source code trying to track through the different function calls etc. all to no avail. But... if you note in the sample.call file, they show how to do Set(CDR(userfield|r)=whatever. I tried adding this |r piece into amaflags, like so: Set(CDR(amaflags|r)=billing Set(CDR(amaflags|r)=omit Set(CDR(amaflags|r)=documentation And it actually works! At least for 1.4beta3 which happens to be running on this box. Regards, Scott From: Scott Keagy Sent: Monday, January 08, 2007 1:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] AMAFlags alwaysDocumentation (or 3 inastcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill) Already using the CDR(userfield), and overloading it with multiple variables will make some DB operations nastier to work with (I'm a little fuzzy and vague on exactly what... something to do with sql joins?). I'm digging deeper into how much pain it really causes us on the DB/App side to see if our work-arounds there are less painful than trying to get AMAFlags to work. It seems that AMAFlags are a really seldom used feature, considering that I'm asking same question that went unanswered several years ago. Thanks all, Scott From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Schøpzinsky Sent: Sunday, January 07, 2007 9:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] AMAFlags alwaysDocumentation (or 3 inastcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill) Hello Why not use the CDR(userfield) field instead. You can set that to any integer of your liking, and use that to identify the type of call. Jon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Keagy Sent: 8. januar 2007 06:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] AMAFlags alwaysDocumentation (or 3 in astcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill) Is anyone out there using AMAFlags? I'd like to set this field as a marker to distinguish different types of calls in CDRs, but can't seem to make it respond to the documented commands Set(CDR(amaflags)=bill) or SetAMAFlags(bill). I've googled this issue, seen others have had this problem with IAX, with different DB drivers for CDR records, etc. I'm using SIP and LOCAL channels, asterisk-1.4beta2 release (I don't think upgrading to current release will fix this problem, it's been around for years based on trouble reports), both text .csv and mysql astcdr.cdr types. Seems like a problem with basic AMAflags support in CDR. They always show up as DOCUMENTATION in the .csv text file, and they always show up as '3' in mysql. I hurt my brain trying to follow the layers of indirection in the source code for where this is actually set. With verbosity turned on in asterisk console I can see the SetAMAFlags function being run. Any tips, tricks, or pointers in the right direction? Thanks, Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] AMAFlags alwaysDocumentation (or 3 inastcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill)
Already using the CDR(userfield), and overloading it with multiple variables will make some DB operations nastier to work with (I'm a little fuzzy and vague on exactly what... something to do with sql joins?). I'm digging deeper into how much pain it really causes us on the DB/App side to see if our work-arounds there are less painful than trying to get AMAFlags to work. It seems that AMAFlags are a really seldom used feature, considering that I'm asking same question that went unanswered several years ago. Thanks all, Scott From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Schøpzinsky Sent: Sunday, January 07, 2007 9:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] AMAFlags alwaysDocumentation (or 3 inastcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill) Hello Why not use the CDR(userfield) field instead. You can set that to any integer of your liking, and use that to identify the type of call. Jon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Keagy Sent: 8. januar 2007 06:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] AMAFlags alwaysDocumentation (or 3 in astcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill) Is anyone out there using AMAFlags? I'd like to set this field as a marker to distinguish different types of calls in CDRs, but can't seem to make it respond to the documented commands Set(CDR(amaflags)=bill) or SetAMAFlags(bill). I've googled this issue, seen others have had this problem with IAX, with different DB drivers for CDR records, etc. I'm using SIP and LOCAL channels, asterisk-1.4beta2 release (I don't think upgrading to current release will fix this problem, it's been around for years based on trouble reports), both text .csv and mysql astcdr.cdr types. Seems like a problem with basic AMAflags support in CDR. They always show up as DOCUMENTATION in the .csv text file, and they always show up as '3' in mysql. I hurt my brain trying to follow the layers of indirection in the source code for where this is actually set. With verbosity turned on in asterisk console I can see the SetAMAFlags function being run. Any tips, tricks, or pointers in the right direction? Thanks, Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMAFlags alwaysDocumentation (or 3 in astcdr mysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill)
Is anyone out there using AMAFlags? I'd like to set this field as a marker to distinguish different types of calls in CDRs, but can't seem to make it respond to the documented commands Set(CDR(amaflags)=bill) or SetAMAFlags(bill). I've googled this issue, seen others have had this problem with IAX, with different DB drivers for CDR records, etc. I'm using SIP and LOCAL channels, asterisk-1.4beta2 release (I don't think upgrading to current release will fix this problem, it's been around for years based on trouble reports), both text .csv and mysql astcdr.cdr types. Seems like a problem with basic AMAflags support in CDR. They always show up as DOCUMENTATION in the .csv text file, and they always show up as '3' in mysql. I hurt my brain trying to follow the layers of indirection in the source code for where this is actually set. With verbosity turned on in asterisk console I can see the SetAMAFlags function being run. Any tips, tricks, or pointers in the right direction? Thanks, Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] centos4.4 x86_64 and zaptel-1.2.12 compile problems?
Anyone seen this and know how to fix it? (note the Assembler messages at
the end). Thanks in advance:
server# make linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o
gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits tones.h
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWAREmakefw.c -o
makefw
./makefw tormenta2.rbt tor2fw tor2fw.h
Loaded 69900 bytes from file
./makefw pciradio.rbt radfw radfw.h
Loaded 42096 bytes from file
ZAPTELVERSION=1.2.12 build_tools/make_version_h version.h.tmp
if cmp -s version.h.tmp version.h ; then echo; else \
mv version.h.tmp version.h ; \
fi
rm -f version.h.tmp
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o ztcfg.o
ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64
-DBUILDING_TONEZONE -o zonedata.lo zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64
-DBUILDING_TONEZONE -o tonezone.lo tonezone.c
ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o libtonezone.a -lm
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o
torisatool.o torisatool.c
cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o
ztmonitor.o ztmonitor.c
cc -o ztmonitor ztmonitor.o
cc -o ztspeed.o -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o zttool.o
zttool.c
cc -o zttool zttool.o -lnewt
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWAREzttest.c -o
zttest
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o
fxotune.o fxotune.c
cc -o fxotune fxotune.o -lm
/lib/modules/2.6.9-42.ELsmp/build
make -C /lib/modules/2.6.9-42.ELsmp/build SUBDIRS=/usr/src/zaptel-1.2.12
modules
make[1]: Entering directory `/usr/src/kernels/2.6.9-42.EL-smp-x86_64'
CC [M] /usr/src/zaptel-1.2.12/zaptel.o
{standard input}: Assembler messages:
{standard input}:16281: Error: suffix or operands invalid for `mov'
{standard input}:16282: Error: suffix or operands invalid for `mov'
{standard input}:16773: Error: suffix or operands invalid for `mov'
{standard input}:16774: Error: suffix or operands invalid for `mov'
{standard input}:17321: Error: suffix or operands invalid for `mov'
{standard input}:17322: Error: suffix or operands invalid for `mov'
{standard input}:17810: Error: suffix or operands invalid for `mov'
{standard input}:17811: Error: suffix or operands invalid for `mov'
/usr/src/zaptel-1.2.12/zaptel.c:188: warning: 'fcstab' defined but not
used
make[2]: *** [/usr/src/zaptel-1.2.12/zaptel.o] Error 1
make[1]: *** [_module_/usr/src/zaptel-1.2.12] Error 2
make[1]: Leaving directory `/usr/src/kernels/2.6.9-42.EL-smp-x86_64'
make: *** [linux26] Error 2
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RE: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14
Henry is my newest Hero :) I'll coordinate with you directly on the releases. Thank you. Regards, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry.L.Coleman Sent: Monday, December 04, 2006 4:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14 Hi Scott, I have the following firmware 1.1.0.16 1.1.0.11 1.1.1.9 1.1.1.14 1.1.2.6 1.1.2.13 Some of these were not from the official website but they were all an improvement 1.1.2.13 is very stable apart from the 56 button ext, unit support. Let me know which ones you want and I can send them to you. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Thanks for your help Claudemir, I look forward to the response. Seems odd that they don't post an archive of their old firmware versions on their website, or at least ones that are required to get to the latest release from whatever is in the field already. Regards, Scott From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claudemir F. Martins Sent: Saturday, December 02, 2006 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14 Hi Scott, I have direct contact with a support person from Grandstream. I will ask him about that and tell you what did he say as soon as possible. Please just wait. Regards Claudemir On 11/30/06, Scott Keagy [EMAIL PROTECTED] wrote: So I've got phones with ancient firmware, and the release notes for 1.1.1.14 say read the previous release notes and first upgrade to 1.1.0.16 The 1.1.0.16 firmware is not available for download from the grandstream website (at least I haven't found it). Any pointers on where to get this intermediate image? I already tried googling to no avail (didn't help that I was using a link with 2000 ms latency). Plus, any overall pointers for making this upgrade process a success would be appreciated. Regards, Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] mwi for voicemail not showing up for realtimeconfig.
A while back I posted a fully functional though somewhat elaborate mechanism to get MWI working with real-time voicemail and NOT using static (static kinda takes a big chunk of value away from real-time). Search the digium Asterisk User forums for my username skeagy with keyword mwi. It does not rely on the built-in sip mechanism. It's a system of scripts that are either triggered by asterisk or a cron-job every one minute to clean out a spool directory, and it uses a uses a template SIP message in a file along with sipsak. It's been working 100% flawlessly in production for 11 months. I'm sure it would work with Asterisk 1.4beta3 assuming that voicemail.conf can still trigger an external script. Regards, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber Sent: Monday, December 04, 2006 4:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. Since I started using 1.4 I'm also not getting MWI. I am not using realtime. MARK. Benjamin Jacob wrote: Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the db. Any ideas, wot else shud i check for? TiA. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] mwi for voicemail not showing up for realtimeconfig.
Here's a link to it: http://forums.digium.com/viewtopic.php?t=4363highlight= Regards, Scott -Original Message- From: Scott Keagy Sent: Monday, December 04, 2006 5:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. A while back I posted a fully functional though somewhat elaborate mechanism to get MWI working with real-time voicemail and NOT using static (static kinda takes a big chunk of value away from real-time). Search the digium Asterisk User forums for my username skeagy with keyword mwi. It does not rely on the built-in sip mechanism. It's a system of scripts that are either triggered by asterisk or a cron-job every one minute to clean out a spool directory, and it uses a uses a template SIP message in a file along with sipsak. It's been working 100% flawlessly in production for 11 months. I'm sure it would work with Asterisk 1.4beta3 assuming that voicemail.conf can still trigger an external script. Regards, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber Sent: Monday, December 04, 2006 4:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. Since I started using 1.4 I'm also not getting MWI. I am not using realtime. MARK. Benjamin Jacob wrote: Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the db. Any ideas, wot else shud i check for? TiA. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Digium TE407P vs. Sangoma A104d
I'd like to hear the feedback too... specific comparison interests: * echo cancellation effectiveness (without introducing artifacts) * ease of provisioning * quality of wiki other docs for giving detailed guidance/hints * customer support from vendor when you encounter problems * PRI protocol variant and IE support (e.g. name display) * Q.SIG support for path replacement * CAS protocol support (EM winkstart? Feature-group D to get ANI?) * load placed on host * resilience to overheating (e.g. in poorly cooled wiring closets) * Heat/BTU output Thanks, Scott From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Monday, December 04, 2006 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Digium TE407P vs. Sangoma A104d Has anyone had experience with one or both of these cards? I'm in a position where I might need to recommend one over the other. I've read everything that I can find online, so now I'd like to hear of personal experiences. Everything I read on both cards is 5 stars! Awesome! It Rocks! They both seem to have similar capabilities, similar pricing, etc. Could those of you who have seen these in action please give us some feedback? I'm interested in anything that might help me decide, be it warranty info, vendor responsiveness, etc. Thanks! -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14
Thanks for your help Claudemir, I look forward to the response. Seems odd that they don't post an archive of their old firmware versions on their website, or at least ones that are required to get to the latest release from whatever is in the field already. Regards, Scott From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claudemir F. Martins Sent: Saturday, December 02, 2006 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14 Hi Scott, I have direct contact with a support person from Grandstream. I will ask him about that and tell you what did he say as soon as possible. Please just wait. Regards Claudemir On 11/30/06, Scott Keagy [EMAIL PROTECTED] wrote: So I've got phones with ancient firmware, and the release notes for 1.1.1.14 say read the previous release notes and first upgrade to 1.1.0.16 The 1.1.0.16 firmware is not available for download from the grandstream website (at least I haven't found it). Any pointers on where to get this intermediate image? I already tried googling to no avail (didn't help that I was using a link with 2000 ms latency). Plus, any overall pointers for making this upgrade process a success would be appreciated. Regards, Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to 1.1.1.14
So I've got phones with ancient firmware, and the release notes for 1.1.1.14 say read the previous release notes and first upgrade to 1.1.0.16 The 1.1.0.16 firmware is not available for download from the grandstream website (at least I haven't found it). Any pointers on where to get this intermediate image? I already tried googling to no avail (didn't help that I was using a link with 2000 ms latency). Plus, any overall pointers for making this upgrade process a success would be appreciated. Regards, Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] voicemail.conf locking problem
If you have enough users where this comes up as a real issue, I'd recommend migrating to Asterisk Realtime voicemail, then can have row-level locking etc. if you use the right kind of storage engine... I've found problems using the dial-by-name directory with realtime voicemail, but it seems you might have the scale where some customization work can be justified. Regards, Scott From: [EMAIL PROTECTED] on behalf of je . Sent: Wed 11/29/2006 4:42 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] voicemail.conf locking problem I'm wondering if anyone is having problems when multiple users concurrently change their voicemail passwords. Consider the following scenario (based on vm_change_password() in app_voicemail.c): - user1 wishes to change his password so voicemail.conf is opened and read into a buffer - user1 changes his password - user2 wishes to change his password so voicemail.conf is opened and read into a buffer - voicemail.conf is written with user1's modified password - voicemail.conf is rewritten with user2's modified password but not including user1's modified password because the voicemail.conf that was read by Asterisk when user2 wanted to change his password was read before the changed password of user1 got written back. It seems by looking at the code that this is how it is currently done. The file is not locked down once it is opened. So my question, is the above scenario correct or is there somewhere a lock which I missed out on? Jez Want to start your own business? Learn how on Yahoo! Small Business. http://smallbusiness.yahoo.com/r-index ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] FW: CISCO 7960G Asterisk
Aww, come on... not everybody has been here for ages or read through years of digests Try the voip-info WIKI: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx Regards, Scott From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Hackensack Sent: Tuesday, November 21, 2006 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: CISCO 7960G Asterisk I was wondering if people have experienced issues with Cisco 7960G and Asterisk. Any feedback on people's experience deploying this phone in production environments would be appreciated. Can you at least do a search first? This same question has been asked so many times that it's been asked to death. Try something different, try Google. Sponsored Link Mortgage rates near 39yr lows. $510,000 Mortgage for $1,698/mo - Calculate new house payment http://www.lowermybills.com/lre/index.jsp?sourceid=lmb-9134-16416moid= 4119 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco media gateways in general
I've never used Asterisk MGCP, and I've only used MGCP gateway on Cisco IOS when controlled from Cisco CallManager (with PRI D-channels backhauled to CallManager). In terms of making an invalid number dialed via Asterisk to Cisco... behavior on Cisco side is entirely subject to how you've programmed the router. If you have no matching dialplan entry, then router will reject the call. If you put a catch-call type of dial peer (i.e. with a destination pattern that matches everything under the sun) and point it out a PSTN connection, then if the call is set up and plays an announcement back to the router with a please hang up and try again type of message, then it may be played back in-band in the audio stream if the router interprets it as an answered call, or it may reject the call toward Asterisk if the PSTN call leg is never established. I suspect this only works as you described when using analog PSTN connections to Cisco gateway, because the loop has to be closed to play back the audio announcement. If you had a PRI on the Cisco gateway, you'd probably get a reject message from the telco and send back a similar reject message toward Asterisk, even if you had the dial peer pointing toward the PSTN for the invalid number. Regards, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek Sent: Wednesday, November 22, 2006 8:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco media gateways in general is possible to control ci$co gateway from asterisk via mgcp? i.e. asterisk as mgcp call agent? PJ Bas van der Veen wrote: Scott, Thanks for the reply. I am experiencing the following with a 2801: - user mistypes a phone number, so the number becomes non-existent - asterisk sends the call to the cisco - the cisco 2801 tries to connect to the non-existent number - the cisco sends a SIP 404 error to asterisk and the call is terminated This behaviour in itself is not weird, but the 2651 and 2821 routers at other branch offices for the same customer DO connect the user to the PSTN and they'd hear a message from the PSTN provider like this number is not in use. I'd like that with the 2801 as well. Would you happen to have the possibility to dial a non-existent number on this setup you mentioned and let me know what the result is? Regards, Bas On Tue, Nov 21, 2006 at 12:03:19PM -0500, Scott Keagy wrote: In my last job I set up Cisco 3845 with PRI cards, talking SIP to Asterisk. No problems there... main trick to get it working for me was to make sure Asterisk was not doing any authentication... add this to a line of the [peer] setup in sip.conf file on Asterisk: insecure=invite,port In terms of IOS side, if you are familiar with enabling sip UA and setting up dial peers, there is nothing special. Regards, Scott From: [EMAIL PROTECTED] on behalf of Bas van der Veen Sent: Tue 11/21/2006 10:02 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco media gateways in general Greetings, After the 0 respones I had on my previous mail regarding the Cisco 2801, I thought I'd be more general. Is anybody using Cisco media gateways at all? If so, how is it working for you? -- Kind regards, Meilleures salutations, Bas van der Veen GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x9E890160 The question of whether a computer can think is no more interesting than the question of whether a submarine can swim. --Edsger Dijkstra ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voice quality of Aastra 480i CT and cordless
Hi Folks, Looking for feedback on the cordless phones with the Aastra 480i CT handset. Is voice quality comparable to standard consumer residential 2.4GHz cordless phones in the US$30 - $50 price range, or better/worse? How about handset and speakerphone quality for the main phone? Seems like there have been various big issues with firmware in past, but is it pretty stable now? Thanks, Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] loosening voicemail file permissions for msg????.txt and msg????.wav
HI folks, I figured out where in the source code to hack the .wav file permissions which were set too restrictive for me, but I cant figure out how to do the same for the .txt file. Looks like the voicemail.c file sets it nicely for asterisk1.4beta3 using a #define statement early on, but msg.txt comes out with permissions 0600 and there are no umask entries that affect how asterisk is started (if anything it would be 022). Ive grepped through the entire source tree from the expanded tarball, and changed every place where it says 0600 to 0666 and recompiled, but still no luck. Ive grepped for umask entries like 077 that might cause the problem, but again no luck. My work-around right now is a cron-job that chmods these directories every minute, but this is ridiculous. Anyone solved this? Thanks, Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue
Title: Re: [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI,not PRI to SIP Authentication Issue When all else fails I resort to adding this in the sip.conf peer config: Insecure=invite,port It took me a while to figure out they can be used together. Regards, Scott - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Sent: Tue Nov 07 15:23:26 2006 Subject: [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI,not PRI to SIP Authentication Issue Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip sip TNT pri pri asterisk The TNT is running 11.0.6 and the asterisk servers are running 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to asterisk but not the other way. The call from asterisk to pri to tnt is good, the TNT is passing SIP invite to the SIP Asterisk server. I have tried many variations of using sip options insecure, autocreatepeer, permit/deny, host, user, etc but can't seem to get asterisk to accept an unauthenticated call from the TNT using SIP. I keep getting SIP/2.0 407 Proxy Authentication Required. I know others have done this, but with older Asterisk versions, I'm wondering what versions of Asterisk are known to work with the MAX TNT and with what version of the TNT? I'm confident this is an asterisk issue, with insecure=very, I should be able to pass calls to asterisk without trying to authenticate it first. But this is not happening. Here is a debug of a call and a snip from my sip.conf: sip.conf [maxtnt] type=friend host=10.10.14.131 insecure=very dtmfmode=inband callerid=MaxTNT maxtnt context=trunktntin qualify=yes reinvite=no canreinvite=no disallow=all allow=ulaw debug lab1*CLI -- SIP read from 10.10.14.131:5060: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 t: sip:[EMAIL PROTECTED]:5060;user=phone f: NO CID NAME sip:[EMAIL PROTECTED]:5060;user=phone;tag=5fe9f589-1fb1f65c-830e0a0a Remote-Party-Id: NO CID NAME sip:[EMAIL PROTECTED]:5060;user=phone;screen=no;id-type=subscriber;party=calling;privacy=off i: [EMAIL PROTECTED] CSeq: 639089 INVITE v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d Max-Forwards: 70 m: sip:[EMAIL PROTECTED]:5060;user=phone k: replaces c: application/sdp Accept: application/sdp Accept-Encoding: Accept-Language: en User-Agent: Lucent-Universal-Gateway l: 232 v=0 o=t1gw01 531756636 531756636 IN IP4 10.10.14.131 s=Session SDP c=IN IP4 10.10.14.131 t=0 0 m=audio 40198 RTP/AVP 0 96 a=silenceSupp:on a=ecan:b on g168 a=ptime:20 a=rtpmap:96 telephone-event/8000 a=rtpmap:0 PCMU/8000 --- (16 headers 11 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.10.14.131 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.10.14.131:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d;received=10.10.14.131 From: NO CID NAME sip:[EMAIL PROTECTED]:5060;user=phone;tag=5fe9f589-1fb1f65c-830e0a0a To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as41f8454e Call-ID: [EMAIL PROTECTED] CSeq: 639089 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=3ea7e98a Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '1239' lab1*CLI -- SIP read from 10.10.14.131:5060: ACK sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 t: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as41f8454e f: NO CID NAME sip:[EMAIL PROTECTED]:5060;user=phone;tag=5fe9f589-1fb1f65c-830e0a0a i: [EMAIL PROTECTED] CSeq: 639089 ACK v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d Max-Forwards: 70 User-Agent: Lucent-Universal-Gateway l: 0 Any guidance will be much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PRI issues
I'm assuming the drop was caused by IP phones renewing leases when the lease expired (and not handling traffic during the renewal period). What kind of IP phones were you using? If my assumptions are off, please clarify the role of DHCP in this issue. Thanks, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Saturday, November 04, 2006 7:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI issues Doug Lytle wrote: Hey everybody, I've, within the last 3 weeks, moved over to a PRI from SBC/ATT. I've received several complaints about dropped calls. Reviewing the archives on PRI and dropped calls shows that I should set the resetinterval=never in the zapata.conf and restart. This Just as a follow up on this. It would appear my problem was caused by having to short of a lease time with DHCP (2 hours). I moved the lease renew time to once every 30 days and for two weeks now, no reports of dropped calls. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
