issue with codec negotiation

[asterisk_at_my_risk]

hello

we are facing a very strange issue with codec negotiation

1)our one to one dialing is working fine there is no issues in that

2) in our auto dialing the calling is getting thru the call is even not getting disconnected but the voice is not flowing and i get follwing error messgage

***********************************************************
Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/3003-081bbeb0 was answered.
-- Executing MeetMe("SIP/3003-081bbeb0", "8600051") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
== Manager 'sendcron' logged off from 127.0.0.1
-- Playing 'conf-onlyperson' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/9919911595825@default-ac06,2", "call_log.agi|9919911 595825") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial("Local/9919911595825@default-ac06,2", "sip/919911595825@SI PTRUNK|55|o") in new stack
-- Called 919911595825@SIPTRUNK
-- SIP/SIPTRUNK-081b9290 is making progress passing it to Local/991991159582 5@default-ac06,2
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/SIPTRUNK-081b9290 answered Local/9919911595825@default-ac06,2
> Channel Local/9919911595825@default-ac06,1 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing AGI("Local/9919911595825@default-ac06,1", "call_log.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
== Spawn extension (default, 9919911595825, 2) exited non-zero on 'Local/99199 11595825@default-ac06,2'
-- Executing DeadAGI("Local/9919911595825@default-ac06,2", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing AGI("SIP/SIPTRUNK-081b9290", "agi-VDADtransfer.agi|8365") in ne w stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("Local/9919911595825@default-ac06,2", "VD_hangup.agi|PR I-----NODEBUG-----16-----ANSWER-----11-----0") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing MeetMe("SIP/SIPTRUNK-081b9290", "8600051") in new stack
-- AGI Script VD_hangup.agi completed, returning 0
it fram e type 64, while native formats is 256 (read/write = 64/64)
[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
le native formats is 256 (read/write = 64/64)
sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
e type 64, while native formats is 256 (read/write = 64/64)
sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:13 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:14 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:14 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:14 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:14 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:14 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:14 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:14 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:14 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:14 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
e type 64, while native formats is 256 (read/write = 64/64)
ound
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/SIPTRUNK-081b 9290'
-- Executing DeadAGI("SIP/SIPTRUNK-081b9290", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("SIP/SIPTRUNK-081b9290", "VD_hangup.agi|PRI-----NODEBUG -----0---------------") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/9919911595825@default-9278,2", "call_log.agi|9919911 595825") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial("Local/9919911595825@default-9278,2", "sip/919911595825@SI PTRUNK|55|o") in new stack
-- Called 919911595825@SIPTRUNK
-- Hungup 'Zap/pseudo-1874516159'
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/3003-081bbeb0 '
-- Executing DeadAGI("SIP/3003-081bbeb0", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("SIP/3003-081bbeb0", "VD_hangup.agi|PRI-----NODEBUG---- -16---------------") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/SIPTRUNK-081b9290 is ringing
-- SIP/SIPTRUNK-081b9290 is making progress passing it to Local/9919911595825@default-9278,2
Oct 27 19:51:33 NOTICE[3827]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 72.37.161.230
-- Got SIP response 486 "EndedByUserBusy-17" back from 85.90.227.72
-- SIP/SIPTRUNK-081b9290 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing Hangup("Local/9919911595825@default-9278,2", "") in new stack
== Spawn extension (default, 9919911595825, 3) exited non-zero on 'Local/9919911595825@default-9278,2'
-- Executing DeadAGI("Local/9919911595825@default-9278,2", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("Local/9919911595825@default-9278,2", "VD_hangup.agi|PRI-----NODEBUG-----17-----BUSY----------") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1

********************************************************

please help us why the call is not getting transfered in auto mode

[asterisk_at_my_risk]

i am using ztdummy with asterisk-1.2.12.1 and astguiclient_2.0.1
also i am using g729 which seems to be working fine because i can call
perfectly in one to one mode

[asterisk_at_my_risk]

i just check g729 codec seem to we working fine
please advice me on this wither i am right or not


localhost*CLI> show translation
Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)

g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc
g723 - - - - - - - - - - -
gsm - - 2 2 2 2 1 4 10 - 15
ulaw - 3 - 1 2 2 1 4 10 - 15
alaw - 3 1 - 2 2 1 4 10 - 15
g726 - 3 2 2 - 2 1 4 10 - 15
adpcm - 3 2 2 2 - 1 4 10 - 15
slin - 2 1 1 1 1 - 3 9 - 14
lpc10 - 4 3 3 3 3 2 - 11 - 16
g729 - 4 3 3 3 3 2 5 - - 16 speex - - - - - - - - - - -
ilbc - 4 3 3 3 3 2 5 11 - -
localhost*CLI>

[asterisk_at_my_risk]

hello Matt
i have treid lots of option from my end but i was unable to find anything on it please
if possible give me some hint on this
this issue is only coming when i am dialing auto calls not in one to one calls

[mflorell]

I don't really know what to tell you, I only use GSM and ULAW codecs and do not have much experience with G729.

[asterisk_at_my_risk]

if we are able to reslove this issue we will surely put its solution on forum

[vctor]

Is your provider supporting g729?

[asterisk_at_my_risk]

yes my vendor is supporting g729 for sure i am able to make preview dialing with no issues on the same vendor with same asterisk
the interisting thing to note here is that i also tried this thing with differen vendors

but error is same

Executing DeadAGI("Local/9919911595825@default-ac06,2", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing AGI("SIP/SIPTRUNK-081b9290", "agi-VDADtransfer.agi|8365") in ne w stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("Local/9919911595825@default-ac06,2", "VD_hangup.agi|PR I-----NODEBUG-----16-----ANSWER-----11-----0") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing MeetMe("SIP/SIPTRUNK-081b9290", "8600051") in new stack
-- AGI Script VD_hangup.agi completed, returning 0
it fram e type 64, while native formats is 256 (read/write = 64/64)
[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
le native formats is 256 (read/write = 64/64)
sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
e type 64, while native formats is 256 (read/write = 64/64)
sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)
Oct 27 19:51:12 WARNING[3766]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)

[rudra_ach]

I think I was having the same problem,But resolved it.
paste your extention.conf trunk configuration for the provider.

rudra

[asterisk_at_my_risk]

hello


thanks for your reply i am realy struck on this

please have a look on it

lemme know if you need any other info from my end

; /etc/asterisk/meetme.conf:
; This is the sample meetme conferencing configuration file

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=Zap/g1 ; Trunk interface
TRUNKX=Zap/g2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com ; IAX trunk interface
SIPTRUNK=SIP/202.88.185.232@202.18.180.72 ; SIP TRUNK


[default]
; Extension 8600 + 8601 conference rooms
exten => 8600,1,Meetme,8600
exten => 8601,1,Meetme,8601


; Extension 3003 for 1st port of the ATA
exten => 3003,1,Playback,transfer|skip
exten => 3003,2,Dial,sip/3003|20|to
exten => 3003,3,Voicemail,3003

; Extension 3004 for 2nd port of the ATA

exten => 3004,1,Playback,transfer|skip
exten => 3004,2,Dial,sip/3004|20|to
exten => 3004,3,Voicemail,3004


; Extension 3004 for 2nd port of the ATA

exten => 3005,1,Playback,transfer|skip
exten => 3005,2,Dial,sip/3005|20|to
exten => 3005,3,Voicemail,3005

; Extension 3004 for 2nd port of the ATA

exten => 3006,1,Playback,transfer|skip
exten => 3006,2,Dial,sip/3006|20|to
exten => 3006,3,Voicemail,3006

; # timeout invalid rules
exten => #,1,Playback(invalid) ; "Thanks for trying the demo"
exten => #,2,Hangup ; Hang them up.
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"

; Give voicemail at extension 8500
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)

; ASTERISK AGENTS LOGINS FOR QUEUES (NOT part of VICIDIAL)
; the following assumes phone agent login and exten are 3 digits and the same
; also assumes that 3-digit login is present in agents.conf and queueus.conf
;Agent Logout then stay onhook, DIAL 54 + 3-digit ID
exten => _54XXX,1,AgentCallbackLogin(||)
; the following are used to login and logout of Asterisk Queues from phone
;Agent Login then stay offhook on the phone, DIAL 55 + 3-digit ID
exten => _55XXX,1,AgentLogin(${EXTEN:1})
;Agent Login then stay onhook, phones will ring, DIAL 56 + 3-digit ID
exten => _56XXX,1,AgentCallbackLogin(||${EXTEN:1}@default)

; Extension 4001 rings Zap phone
exten => 4001,1,Dial,Zap/1|30| ; ring Zap device 1
exten => 4001,2,Voicemail,u4001 ; Send to voicemail...

exten => h,1,DeadAGI(call_log.agi,${EXTEN}) ; DeadAGI is new
exten => h,2,DeadAGI(VD_hangup.agi,PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; Extension 3429 - Inbound 800 number (1-800-555-3429)
exten => _**3429,1,Ringing
exten => _**3429,2,AGI(call_log.agi,${EXTEN})
exten => _**3429,3,Answer
exten => _**3429,4,Dial,sip/spa2000&sip/spa2001|30|to
exten => _**3429,5,Voicemail,u2000
; Extension 3429 - with ANI [callerID]
exten => _*NXXNXXXXXX*3429,1,Ringing
exten => _*NXXNXXXXXX*3429,2,AGI(call_log.agi,${EXTEN})
exten => _*NXXNXXXXXX*3429,3,Answer
exten => _*NXXNXXXXXX*3429,4,Dial,sip/spa2000&sip/spa2001|30|to
exten => _*NXXNXXXXXX*3429,5,Voicemail,u2000

; Inbound call from BINFONE
; exten => 1112223333,1,AGI(call_log.agi,${EXTEN})
; exten => 1112223333,2,Dial(sip/gs102,55,o)
; exten => 1112223333,3,Hangup

; Extension 7275551212 - Inbound local number from PRI with 10 digit delivery
exten => 7275551212,1,Ringing
exten => 7275551212,2,Wait(1)
exten => 7275551212,3,AGI(call_logCID.agi,${EXTEN}-----${CALLERID}-----${CALLERIDNUM}-----${CALLERIDNAME})
exten => 7275551212,4,Answer
exten => 7275551212,5,Dial,sip/spa2000&sip/spa2001|30|to
exten => 7275551212,6,Voicemail,u2000

; dial a long distance outbound number to the INDIA
exten => _991XXXXXXXXXX,1,AGI(call_log.agi,${EXTEN})
;exten => _991XXXXXXXXXX,2,Dial(${SIPTRUNK}/${EXTEN:1},55,tTo)
exten => _991XXXXXXXXXX,2,Dial(sip/${EXTEN:1}@SIPTRUNK,55,o)
exten => _991XXXXXXXXXX,3,Hangup

; dial a long distance outbound number to Australia
exten => _901161XXXXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _901161XXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,tTo)
exten => _901161XXXXXXXXX,3,Hangup



; dial a long distance outbound number
exten => _97651NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _97651NXXNXXXXXX,2,Dial(sip/${EXTEN:1}@SIPTRUNK,55,o)
exten => _97651NXXNXXXXXX,3,Hangup
; dial a long distance outbound number through BINFONE
; exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
; exten => _91NXXNXXXXXX,2,Dial(${TRUNKIAX}/${EXTEN},55,o)
; exten => _91NXXNXXXXXX,3,Hangup
; dial a long distance outbound number through a SIP provider
; exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
; exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN}@SIPtrunk,55,o)
; exten => _91NXXNXXXXXX,3,Hangup

; dial a local outbound number (modified because of only LD T1)
;exten => _9NXXXXXX,1,AGI(call_log.agi,${EXTEN})
;exten => _9NXXXXXX,2,Dial(${TRUNK}/1727${EXTEN:1},,tTo)
;exten => _9NXXXXXX,3,Hangup

; parameters for call_inbound.agi (7 fields separated by five dashes "-----"):
; 1. the extension of the phone to ring as defined in the asterisk.phones table
; 2. the phone number that was called, for the live_inbound/_log entry
; 3. a text description of the number that was called in
; 4-7. optional fields, they are also passed as fields in the GUI to web browser

; Extension 3429 - Inbound 800 number (1-800-555-3429)
exten => _**3429,1,Ringing
exten => _**3429,2,AGI(call_log.agi,${EXTEN})
exten => _**3429,3,AGI(call_inbound.agi,spa2000-----8005553429-----Inbound 800-----x-----y-----z-----w)
exten => _**3429,4,Answer
exten => _**3429,5,Dial,sip/spa2000&sip/spa2001|30|to
exten => _**3429,6,Voicemail,u2000
; Extension 3429 - with ANI [callerID]
exten => _*NXXNXXXXXX*3429,1,Ringing
exten => _*NXXNXXXXXX*3429,2,AGI(call_log.agi,${EXTEN})
exten => _*NXXNXXXXXX*3429,3,AGI(call_inbound.agi,spa2000-----8005553429-----Inbound 800-----x-----y-----z-----w)
exten => _*NXXNXXXXXX*3429,3,Answer
exten => _*NXXNXXXXXX*3429,4,Dial,sip/spa2000&sip/spa2001|30|to
exten => _*NXXNXXXXXX*3429,5,Voicemail,u2000

exten => _010*010*010*015*8600XXX,1,Goto(default,${EXTEN:16},1)
exten => _010*010*010*015*8600XXX*.,1,Goto(default,${EXTEN:16},1)
exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)


; ### server 2 extens:
; exten => _010*010*010*016*8600XXX,1,Dial(${TRUNKIAX1}/${EXTEN:16},55,o)
; exten => _010*010*010*016*8600XXX,2,Hangup
; exten => _010*010*010*016*8600XXX*.,1,Dial(${TRUNKIAX1}/${EXTEN:16},55,o)
; exten => _010*010*010*016*8600XXX*.,2,Hangup


; parameters for agi-VDADcloser.agi (2 fields separated by five dashes "-----"):
; 1. the full extension formatted by VICIDIAL for internal transfers * separated
; 2. the word START to denote the beginning of the acceptance of the transfer
; inbound VICIDIAL transfer calls [can arrive through PRI T1 crossover or IAX channel]
exten => _90009.,1,Answer ; Answer the line
exten => _90009.,2,AGI(agi-VDADcloser.agi,${EXTEN}-----START)
exten => _90009.,3,Hangup

; parameters for agi-VDADcloser_inbound....agi (7 fields separated by five dashes "-----"):
; 1. the full name of the IN GROUP to be used in vicidial for the inbound call
; 2. the phone number that was called, for the log entry
; 3. the callerID or lead_id of the person that called(usually overridden)
; 4. the park extension audio file name if used
; 5. the status of the call initially(usually not used)
; 6. the list_id to insert the new lead under if it is new (and callerID available)
; 7. the phone dialing code to insert with the new lead if new (and callerID available)

; inbound VICIDIAL call with CID delivery through T1 PRI
exten => 1234,1,Ringing ; call ringing
exten => 1234,2,Wait(1) ; Wait 1 second for CID delivery from PRI
exten => 1234,3,Answer ; Answer the line
exten => 1234,4,AGI(agi-VDADcloser_inboundCID.agi,CL_GALLERIA-----7275555134-----Closer-----park----------999-----1)
exten => 1234,5,Hangup

; inbound VICIDIAL call with ANI delivery through robbed-bit T1 (*NXXNXXXXXX*DNIS)
exten => 1234,1,Answer ; Answer the line
exten => 1234,2,AGI(agi-VDADcloser_inboundANI.agi,CL_GALLERIA-----7275555134-----Closer-----park----------999-----1)
exten => 1234,3,Hangup

; inbound VICIDIAL call with prompt for 4-digit fronter code
exten => 1234,1,Answer ; Answer the line
exten => 1234,2,AGI(agi-VDADcloser_inbound.agi,CL_GALLERIA-----7275555134-----Closer-----park----------999-----1)
exten => 1234,3,Hangup

; barge monitoring extension
exten => 8159,1,ZapBarge
exten => 8159,2,Hangup

; ZapBarge direct channel extensions
exten => 8612001,1,ZapBarge(1)
exten => 8612002,1,ZapBarge(2)
exten => 8612003,1,ZapBarge(3)
exten => 8612004,1,ZapBarge(4)
exten => 8612005,1,ZapBarge(5)
exten => 8612006,1,ZapBarge(6)
exten => 8612007,1,ZapBarge(7)
exten => 8612008,1,ZapBarge(8)
exten => 8612009,1,ZapBarge(9)
exten => 8612010,1,ZapBarge(10)
exten => 8612011,1,ZapBarge(11)
exten => 8612012,1,ZapBarge(12)
exten => 8612013,1,ZapBarge(13)
exten => 8612014,1,ZapBarge(14)
exten => 8612015,1,ZapBarge(15)
exten => 8612016,1,ZapBarge(16)
exten => 8612017,1,ZapBarge(17)
exten => 8612018,1,ZapBarge(18)
exten => 8612019,1,ZapBarge(19)
exten => 8612020,1,ZapBarge(20)
exten => 8612021,1,ZapBarge(21)
exten => 8612022,1,ZapBarge(22)
exten => 8612023,1,ZapBarge(23)
exten => 8612024,1,ZapBarge(24)

; astGUIclient conferences
exten => 8600011,1,Meetme,8600011|q
exten => 8600012,1,Meetme,8600012|q
exten => 8600013,1,Meetme,8600013|q
exten => 8600014,1,Meetme,8600014|q
exten => 8600015,1,Meetme,8600015|q
exten => 8600016,1,Meetme,8600016|q
exten => 8600017,1,Meetme,8600017|q
exten => 8600018,1,Meetme,8600018|q
exten => 8600019,1,Meetme,8600019|q
exten => 8600020,1,Meetme,8600020|q
exten => 8600021,1,Meetme,8600021|q
exten => 8600022,1,Meetme,8600022|q
exten => 8600023,1,Meetme,8600023|q
exten => 8600024,1,Meetme,8600024|q
exten => 8600025,1,Meetme,8600025|q
exten => 8600026,1,Meetme,8600026|q
exten => 8600027,1,Meetme,8600027|q
exten => 8600028,1,Meetme,8600028|q
exten => 8600029,1,Meetme,8600029|q

; VICIDIAL conferences
exten => 8600051,1,Meetme,8600051
exten => 8600052,1,Meetme,8600052
exten => 8600053,1,Meetme,8600053
exten => 8600054,1,Meetme,8600054
exten => 8600055,1,Meetme,8600055
exten => 8600056,1,Meetme,8600056
exten => 8600057,1,Meetme,8600057
exten => 8600058,1,Meetme,8600058
exten => 8600059,1,Meetme,8600059
exten => 8600060,1,Meetme,8600060
exten => 8600061,1,Meetme,8600061
exten => 8600062,1,Meetme,8600062
exten => 8600063,1,Meetme,8600063
exten => 8600064,1,Meetme,8600064
exten => 8600065,1,Meetme,8600065
exten => 8600066,1,Meetme,8600066
exten => 8600067,1,Meetme,8600067
exten => 8600068,1,Meetme,8600068
exten => 8600069,1,Meetme,8600069
exten => 8600070,1,Meetme,8600070
exten => 8600071,1,Meetme,8600071
exten => 8600072,1,Meetme,8600072
exten => 8600073,1,Meetme,8600073
exten => 8600074,1,Meetme,8600074
exten => 8600075,1,Meetme,8600075
exten => 8600076,1,Meetme,8600076
exten => 8600077,1,Meetme,8600077
exten => 8600078,1,Meetme,8600078
exten => 8600079,1,Meetme,8600079
exten => 8600080,1,Meetme,8600080
exten => 8600081,1,Meetme,8600081
exten => 8600082,1,Meetme,8600082
exten => 8600083,1,Meetme,8600083
exten => 8600084,1,Meetme,8600084
exten => 8600085,1,Meetme,8600085
exten => 8600086,1,Meetme,8600086
exten => 8600087,1,Meetme,8600087
exten => 8600088,1,Meetme,8600088
exten => 8600089,1,Meetme,8600089
exten => 8600090,1,Meetme,8600090
exten => 8600091,1,Meetme,8600091
exten => 8600092,1,Meetme,8600092
exten => 8600093,1,Meetme,8600093
exten => 8600094,1,Meetme,8600094
exten => 8600095,1,Meetme,8600095
exten => 8600096,1,Meetme,8600096
exten => 8600097,1,Meetme,8600097
exten => 8600098,1,Meetme,8600098
exten => 8600099,1,Meetme,8600099
exten => 8600100,1,Meetme,8600100
; quiet entry and leaving conferences for VICIDIAL
exten => 78600051,1,Meetme,8600051|q
exten => 78600052,1,Meetme,8600052|q
exten => 78600053,1,Meetme,8600053|q
exten => 78600054,1,Meetme,8600054|q
exten => 78600055,1,Meetme,8600055|q
exten => 78600056,1,Meetme,8600056|q
exten => 78600057,1,Meetme,8600057|q
exten => 78600058,1,Meetme,8600058|q
exten => 78600059,1,Meetme,8600059|q
exten => 78600060,1,Meetme,8600060|q
exten => 78600061,1,Meetme,8600061|q
exten => 78600062,1,Meetme,8600062|q
exten => 78600063,1,Meetme,8600063|q
exten => 78600064,1,Meetme,8600064|q
exten => 78600065,1,Meetme,8600065|q
exten => 78600066,1,Meetme,8600066|q
exten => 78600067,1,Meetme,8600067|q
exten => 78600068,1,Meetme,8600068|q
exten => 78600069,1,Meetme,8600069|q
exten => 78600070,1,Meetme,8600070|q
exten => 78600071,1,Meetme,8600071|q
exten => 78600072,1,Meetme,8600072|q
exten => 78600073,1,Meetme,8600073|q
exten => 78600074,1,Meetme,8600074|q
exten => 78600075,1,Meetme,8600075|q
exten => 78600076,1,Meetme,8600076|q
exten => 78600077,1,Meetme,8600077|q
exten => 78600078,1,Meetme,8600078|q
exten => 78600079,1,Meetme,8600079|q
exten => 78600080,1,Meetme,8600080|q
exten => 78600081,1,Meetme,8600081|q
exten => 78600082,1,Meetme,8600082|q
exten => 78600083,1,Meetme,8600083|q
exten => 78600084,1,Meetme,8600084|q
exten => 78600085,1,Meetme,8600085|q
exten => 78600086,1,Meetme,8600086|q
exten => 78600087,1,Meetme,8600087|q
exten => 78600088,1,Meetme,8600088|q
exten => 78600089,1,Meetme,8600089|q
exten => 78600090,1,Meetme,8600090|q
exten => 78600091,1,Meetme,8600091|q
exten => 78600092,1,Meetme,8600092|q
exten => 78600093,1,Meetme,8600093|q
exten => 78600094,1,Meetme,8600094|q
exten => 78600095,1,Meetme,8600095|q
exten => 78600096,1,Meetme,8600096|q
exten => 78600097,1,Meetme,8600097|q
exten => 78600098,1,Meetme,8600098|q
exten => 78600099,1,Meetme,8600099|q
exten => 78600100,1,Meetme,8600100|q
; quiet monitor extensions for meetme rooms (for room managers)
exten => 68600051,1,Meetme,8600051|mq
exten => 68600052,1,Meetme,8600052|mq
exten => 68600053,1,Meetme,8600053|mq
exten => 68600054,1,Meetme,8600054|mq
exten => 68600055,1,Meetme,8600055|mq
exten => 68600056,1,Meetme,8600056|mq
exten => 68600057,1,Meetme,8600057|mq
exten => 68600058,1,Meetme,8600058|mq
exten => 68600059,1,Meetme,8600059|mq
exten => 68600060,1,Meetme,8600060|mq
exten => 68600061,1,Meetme,8600061|mq
exten => 68600062,1,Meetme,8600062|mq
exten => 68600063,1,Meetme,8600063|mq
exten => 68600064,1,Meetme,8600064|mq
exten => 68600065,1,Meetme,8600065|mq
exten => 68600066,1,Meetme,8600066|mq
exten => 68600067,1,Meetme,8600067|mq
exten => 68600068,1,Meetme,8600068|mq
exten => 68600069,1,Meetme,8600069|mq
exten => 68600070,1,Meetme,8600070|mq
exten => 68600071,1,Meetme,8600071|mq
exten => 68600072,1,Meetme,8600072|mq
exten => 68600073,1,Meetme,8600073|mq
exten => 68600074,1,Meetme,8600074|mq
exten => 68600075,1,Meetme,8600075|mq
exten => 68600076,1,Meetme,8600076|mq
exten => 68600077,1,Meetme,8600077|mq
exten => 68600078,1,Meetme,8600078|mq
exten => 68600079,1,Meetme,8600079|mq
exten => 68600080,1,Meetme,8600080|mq
exten => 68600081,1,Meetme,8600081|mq
exten => 68600082,1,Meetme,8600082|mq
exten => 68600083,1,Meetme,8600083|mq
exten => 68600084,1,Meetme,8600084|mq
exten => 68600085,1,Meetme,8600085|mq
exten => 68600086,1,Meetme,8600086|mq
exten => 68600087,1,Meetme,8600087|mq
exten => 68600088,1,Meetme,8600088|mq
exten => 68600089,1,Meetme,8600089|mq
exten => 68600090,1,Meetme,8600090|mq
exten => 68600091,1,Meetme,8600091|mq
exten => 68600092,1,Meetme,8600092|mq
exten => 68600093,1,Meetme,8600093|mq
exten => 68600094,1,Meetme,8600094|mq
exten => 68600095,1,Meetme,8600095|mq
exten => 68600096,1,Meetme,8600096|mq
exten => 68600097,1,Meetme,8600097|mq
exten => 68600098,1,Meetme,8600098|mq
exten => 68600099,1,Meetme,8600099|mq
exten => 68600100,1,Meetme,8600100|mq

; park channel for client GUI parking, hangup after 30 minutes
; create a GSM formatted audio file named "park.gsm" that is 30 minutes long
; and put it in /var/lib/asterisk/sounds
exten => 8301,1,Answer
exten => 8301,2,AGI(park_CID.agi)
exten => 8301,3,Playback(park)
exten => 8301,4,Hangup
exten => 8303,1,Answer
exten => 8303,2,AGI(park_CID.agi)
exten => 8303,3,Playback(conf)
exten => 8303,4,Hangup

; park channel for client GUI conferencing, hangup after 30 minutes
; create a GSM formatted audio file named "conf.gsm" that is 30 minutes long
; and put it in /var/lib/asterisk/sounds
exten => 8302,1,Answer
exten => 8302,2,Playback(conf)
exten => 8302,3,Hangup

; default audio for safe harbor 2-second-after-hello message then hangup
; create a GSM formatted audio file complies with safe harbor rules
; and put it in /var/lib/asterisk/sounds then change filename below
exten => 8307,1,Answer
exten => 8307,2,Playback(vm-goodbye)
exten => 8307,3,Hangup

; this is used for recording conference calls, the client app sends the filename
; value as a callerID recordings go to /var/spool/asterisk/monitor (WAV)
exten => 8309,1,Answer
exten => 8309,2,Monitor(wav,${CALLERIDNAME})
exten => 8309,3,Wait,3600
exten => 8309,4,Hangup
; this is used for recording conference calls, the client app sends the filename
; value as a callerID recordings go to /var/spool/asterisk/monitor (GSM)
exten => 8310,1,Answer
exten => 8310,2,Monitor(gsm,${CALLERIDNAME})
exten => 8310,3,Wait,3600
exten => 8310,4,Hangup

; this is used for playing a message to an answering machine forwarded from AMD in VICIDIAL
; replace conf with the message file you want to leave
exten => 8320,1,WaitForSilence(2000,2) ; AMD got machine. leave message after recording
exten => 8320,2,Playback(conf)
exten => 8320,3,AGI(VD_amd_post.agi,${EXTEN})
exten => 8320,4,Hangup

; this is used to allow the GUI to send you directly into voicemail
; don't forget to set GUI variable $voicemail_exten to this extension
exten => 8501,1,VoicemailMain(s${CALLERIDNUM})
exten => 8501,2,Hangup

; this is used to allow the GUI to send live calls directly into voicemail
; don't forget to set GUI variable $voicemail_dump_exten to this extension
exten => _85026666666666.,1,Wait(2)
exten => _85026666666666.,2,Voicemail(${EXTEN:14})
exten => _85026666666666.,3,Hangup

; this is used for sending DTMF signals within conference calls, the client app
; sends the digits to be played in the callerID field
; sound files must be placed in /var/lib/asterisk/sounds
exten => 8500998,1,Answer
exten => 8500998,2,Playback(silence)
exten => 8500998,3,AGI(agi-dtmf.agi)
exten => 8500998,4,Hangup

; prompt recording AGI script, ID is 4321
exten => 8168,1,Answer
exten => 8168,2,AGI(agi-record_prompts.agi)
exten => 8168,3,Hangup

; playback of recorded prompts
exten => _851XXXXX,1,Answer
exten => _851XXXXX,2,Playback(${EXTEN})
exten => _851XXXXX,3,Hangup

; VICIDIAL_auto_dialer transfer script:
exten => 8365,1,AGI(call_log.agi,${EXTEN})
exten => 8365,2,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,3,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,4,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,5,Hangup

; VICIDIAL_auto_dialer transfer script SURVEY at beginning:
exten => 8366,1,AGI(call_log.agi,${EXTEN})
exten => 8366,2,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,3,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,4,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balance Overflow:
exten => 8367,1,AGI(call_log.agi,${EXTEN})
exten => 8367,2,AGI(agi-VDAD_LO_transfer.agi,${EXTEN})
exten => 8367,3,AGI(agi-VDAD_LO_transfer.agi,${EXTEN})
exten => 8367,4,AGI(agi-VDAD_LO_transfer.agi,${EXTEN})
exten => 8367,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,AGI(call_log.agi,${EXTEN})
exten => 8368,2,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8368,3,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8368,4,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8368,5,Hangup

; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten => 8369,1,AGI(call_log.agi,${EXTEN})
exten => 8369,2,AMD(3500|1500|300|5000|120|50|5|256)
exten => 8369,3,AGI(VD_amd.agi,${EXTEN})
exten => 8369,4,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8369,5,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8369,6,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8369,7,Hangup

; VICIDIAL auto-dial reminder script
exten => 8372,1,AGI(call_log.agi,${EXTEN})
exten => 8372,2,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,3,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,4,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,5,Hangup

[asterisk_at_my_risk]

; dial a long distance outbound number to the INDIA
exten => _991XXXXXXXXXX,1,AGI(call_log.agi,${EXTEN})
;exten => _991XXXXXXXXXX,2,Dial(${SIPTRUNK}/${EXTEN:1},55,tTo)
exten => _991XXXXXXXXXX,2,Dial(sip/${EXTEN:1}@SIPTRUNK,55,o)
exten => _991XXXXXXXXXX,3,Hangup

he support g729 i am able to make a call thru him on preview mode(one to one)
but not in automatic dialing

[rudra_ach]

Hi,

; dial a long distance outbound number to the INDIA
exten => _991XXXXXXXXXX,1,AGI(call_log.agi,${EXTEN})
;exten => _991XXXXXXXXXX,2,Dial(${SIPTRUNK}/${EXTEN:1},55,tTo)
exten => _991XXXXXXXXXX,2,Dial(sip/${EXTEN:1}@SIPTRUNK,55,o)
exten => _991XXXXXXXXXX,3,Hangup

from your above script it seems that after the client at the other end picks up the phone asterisk should transfer the call to one extention in vicidial.But here as you are not allowing the transfer after connection of call it goes to deadagi,

My script bellow working fine for me.

exten => _6.,1,AGI(call_log.agi,${EXTEN})
exten => _6.,2,Dial(SIP/username/${EXTEN:1},30,tTo)
exten => _6.,3,Hangup

look at the string at last tTo.

bellow the sip trunk configuration.

[username]
authuser=username
fromdomain=ipaddress
fromuser=username
host=ipaddress
insecure=very
nat=no
qualify=yes
secret=password
sendrpid=yes
type=peer
username=username@ipaddress

Hope this will help

ciao
rudra

; dial a long distance outbound number to the INDIA
exten => _991XXXXXXXXXX,1,AGI(call_log.agi,${EXTEN})
;exten => _991XXXXXXXXXX,2,Dial(${SIPTRUNK}/${EXTEN:1},55,tTo)
exten => _991XXXXXXXXXX,2,Dial(sip/${EXTEN:1}@SIPTRUNK,55,o)
exten => _991XXXXXXXXXX,3,Hangup

he support g729 i am able to make a call thru him on preview mode(one to one)
but not in automatic dialing

[asterisk_at_my_risk]

i treid the option you gave but the error is same

chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64)


can you tell me the reason why this error
comes usualy
that may help to investigate it further

waiting for your response

Regards
RAHUL

[asterisk_at_my_risk]

hello
moderators and freinds i am struck up with a very strange issue related with codec i have worked realy hard on this but without any success please give me some hints if possible so that i can try

i have spent huge time on this issue now any suggestions from any of you are welcome it may help me to think in a fresh way

MY ISSUE IS...........

i am able to call perfectly in manual mode but when i cam ion auto mode i am getting the warning mentioned below
i am using g729

*************************************************************


SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK0913f431;rport=5060
Record-Route: <sip:85.90.227.72;ftag=as6c7b24fb;lr>
From: V1110233033000000007 <sip:0000000000@203.122.26.232>;tag=as6c7b24fb
To: <sip:919811412508@85.90.227.72>;tag=48811405a147396d5056a15585bf817f
Call-ID: 6fa049644021b6210900545873ff6c94@203.122.26.232
CSeq: 102 INVITE
Server: Sippy
Contact: Anonymous <sip:85.90.227.72:5061>
Content-Length: 237
Content-Type: application/sdp

v=0
o=Sippy 145669356 1 IN IP4 85.90.227.72
s=session controller
t=0 0
m=audio 11356 RTP/AVP 18 101
c=IN IP4 72.37.161.230
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (11 headers 11 lines)---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 72.37.161.230:11356
Found description format G729
Found description format telephone-event
Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (n othing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event) , combined - 0x1 (telephone-event)
list_route: hop: <sip:85.90.227.72;ftag=as6c7b24fb;lr>
set_destination: Parsing <sip:85.90.227.72;ftag=as6c7b24fb;lr> for address/port to send to
set_destination: set destination to 85.90.227.72, port 5060
Transmitting (no NAT) to 85.90.227.72:5060:
ACK sip:85.90.227.72:5061 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK04db743d;rport
Route: <sip:85.90.227.72;ftag=as6c7b24fb;lr>
From: "V1110233033000000007" <sip:0000000000@203.122.26.232>;tag=as6c7b24fb
To: <sip:919811412508@85.90.227.72>;tag=48811405a147396d5056a15585bf817f
Contact: <sip:0000000000@203.122.26.232>
Call-ID: 6fa049644021b6210900545873ff6c94@203.122.26.232
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/SIPTRUNK-099cf748 answered Local/9919811412508@default-a609,2
> Channel Local/9919811412508@default-a609,1 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing AGI("Local/9919811412508@default-a609,1", "call_log.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
== Spawn extension (default, 9919811412508, 2) exited non-zero on 'Local/99198 11412508@default-a609,2'
-- Executing DeadAGI("Local/9919811412508@default-a609,2", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing AGI("SIP/SIPTRUNK-099cf748", "agi-VDADtransfer.agi|8365") in ne w stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("Local/9919811412508@default-a609,2", "VD_hangup.agi|PR I-----NODEBUG-----16-----ANSWER-----27-----0") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi

<-- SIP read from 203.122.26.234:61071:


--- (0 headers 0 lines) Nat keepalive ---
Nov 10 23:31:06 NOTICE[12252]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 72.37.161.230


-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- AGI Script VD_hangup.agi completed, returning 0
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[12252]: chan_sip.c:2570 sip_write: Asked to transmit fra me type 64, while native formats is 256 (read/write = 64/64)
Nov 10 23:31:09 WARNING[122

waiting for some reply

[mflorell]

Have you called Digium about this issue with G729?

[Op3r]

I think he is using ztdummy with the free g729

I used to have that before when I use ztdummy with the free g729

then when I used x100p it went away. but still get some errors like that semi flooding my cli. then when I used the licensed one it went away.

asterisk at my risk. type show translation and show g729 on your cli and post the output here and also what zaptel version are you using?

[asterisk_at_my_risk]

thanks Matt for your reply yes i have did that but i have already done that
what he suggested

only thing i am wondering is that if there was something wrong with the codec how could manual calls have worked
and if i try to record calls in manual mode if that is working it should mean that my g729 is working as my vendor is supporting only g729
and






Hello,

By your logs its trying to transcode from sln to g729. In your sip.conf under general add
the following:

disallow=all
allow=g729
allow=ulaw
allow=alaw

Regards,

--
James Heintschel
support@digium.com
+1.256.428.6161
+1.877.LINUX.ME (546.8963)

[asterisk_at_my_risk]

hello moderators

firstly thanks a lot for your reply yes you are right i am using ztdummy
with free g729 but i have replace that free g729 with paid one from digium

here are the outputs
sachitel*CLI> show g729
0/0 encoders/decoders of 8 licensed channels are currently in use

sachitel*CLI> show translation
Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)

g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc
g723 - - - - - - - - - - -
gsm - - 2 2 2 2 1 5 11 33 15
ulaw - 4 - 1 2 2 1 5 11 33 15
alaw - 4 1 - 2 2 1 5 11 33 15
g726 - 4 2 2 - 2 1 5 11 33 15
adpcm - 4 2 2 2 - 1 5 11 33 15
slin - 3 1 1 1 1 - 4 10 32 14
lpc10 - 5 3 3 3 3 2 - 12 34 16
g729 - 5 3 3 3 3 2 6 - 34 16
speex - 5 3 3 3 3 2 6 12 - 16
ilbc - 7 5 5 5 5 4 8 14 36 -
sachitel*CLI>




i will be greatful to know from you followings things

1.if i use free g729 with x100p card
2.or if i use paid g729 without X100p(ie with ztdummy)what option do you recomend me

the only reason i am wary of using digium original one is not the money but in case i need to format my system i will loose my license

waiting for your reply

regards

asterisk_at_my_risk

[Op3r]

You wont lose your license even when you reformat your hard drive. You can always contact digium if you run into troubles about reinstalling your codec license.

when u used the digium codec do you still get the errors flooding your cli?

my best recommendation is

get an x100p and use alaw/ulaw (if you have a very high speed leased line for voip)
or if you have high jitters connecting to your voip provider
x100p with licensed digium codecs.

but I once tried using the free codec for my testing server with x100p and though i get minimal errors like that.

or you can try to experiment like reinstalling you zaptel and asterisk to see if it alleviates the problem.

[asterisk_at_my_risk]

thanks for your reply i will work on this and will get back to you
with results

here is india almost all voip provider prefer to use g729 only
so i have no other option but to use g729 only

[Op3r]

How many hops do you get before connecting reaching your voip provider? do you guys have a lot of jitters in your internet connection there?

you can tell them to accept alaw and ulaw. thats if you only get like 5-8 hops to get to their server. hehehe.

anyway, u havent answered my question yet. do you still have errors flooding your cli when ur using the licensed g729? can you post your loadavg? is that a single server install?

[asterisk_at_my_risk]

Hello


yes i still get these errors when i am having a licensed version
right now i am having a license for 4 channels
here is my loadavg details

VICIDIAL: Realtime Campaign: STOP | GO MODIFY | REPORTS
DIAL LEVEL: 1 FILTER: NONE TIME: 2006-11-18 04:42:13
DIALABLE LEADS: 3 CALLS TODAY: 4 AVG AGENTS: 1 DIAL METHOD: RATIO
HOPPER LEVEL: 10 CALLS DROPPED: 0 DL DIFF: 0.2 STATUSES: NEW, N, INCALL, NA, DROP
LEADS IN HOPPER: 3 DROPPED PERCENT: 0% DIFF: 20.00% ORDER: DOWN
+ VIEW MORE SETTINGS
NO LIVE CALLS WAITING
1 agents logged in 1 agents in calls 0 agents waiting 0 paused agents

VICIDIAL: Agents Time On Calls Campaign: CL 2006-11-18 04:42:13

+------------|--------+-----------+--------+-----------------+-----------------+---------+------------+
| STATION | USER | SESSIONID | STATUS | SERVER IP | CALL SERVER IP | MM:SS | CAMPAIGN |
+------------|--------+-----------+--------+-----------------+-----------------+---------+------------+
| SIP/3003 | 6666 | 8600051 | INCALL | 203.122.26.232 | | 0:42 | CL |
+------------|--------+-----------+--------+-----------------+-----------------+---------+------------+
1 agents logged in on all servers
System Load Average: 0.03

and right now i am on a 2 server

server 1
Mysql/apache/astguiclient

server 2

asterisk/vicidial

Note:right now i am on ztdummy
ms lsmod output
Module Size Used by
ztdummy 3924 0
zaptel 210820 5 ztdummy
crc_ccitt 2113 1 zaptel
md5 4033 1
ipv6 268097 14
parport_pc 28933 1
lp 13001 0


but i have faced the same issue on single server instalation also

please help me on this

waiting for your reply







How many hops do you get before connecting reaching your voip provider? do you guys have a lot of jitters in your internet connection there?

you can tell them to accept alaw and ulaw. thats if you only get like 5-8 hops to get to their server. hehehe.

anyway, u havent answered my question yet. do you still have errors flooding your cli when ur using the licensed g729? can you post your loadavg? is that a single server install?
_________________
Asterisk is cool. Lots of people are using it, why shouldnt you?

[mflorell]

Support is included when you buy the G729 licenses from Digium, have you contacted them about this problem?

[asterisk_at_my_risk]

yes i have contacted digium about this there reply is to upgrade
my asterisk version to 1.2.13 from 1.2.12
which is very strange because from what i know g729 should work for
1.2.12 also

can you tell me what specific thing i shall ask from them

Thanks for your help

[mflorell]

There is a bug with Asterisk 1.2.12 that makes VICIDIAL auto-dialing not work for most VOIP channels, so you need to upgrade to at least 1.2.12.1 for that to be fixed.

[asterisk_at_my_risk]

sir
i am sorry my version is already 1.2.12.1 actualy i forget to include that 1


Connected to Asterisk 1.2.12.1 currently running on Asterisk1 (pid = 2660)
Verbosity is at least 10
Asterisk1*CLI> show g729
0/0 encoders/decoders of 4 licensed channels are currently in use
Asterisk1*CLI>
Asterisk1*CLI>

[ramindia]

hi

you should register the codecs.

Please become familar with the backup procedures listed
at ftp://ftp.digium.com/pub/asterisk/g729/README. In addition, please
read our G.729 policy which is available at
http://www.digium.com/en/products/voice/g729policy.php .


should help you

let me know if it works

Ram

[asterisk_at_my_risk]

hello
i have taken a backup of my codec and as far as registration is consider
i fell that my codec are registered
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.12.1 currently running on Asterisk1 (pid = 2660)
Verbosity is at least 10
Asterisk1*CLI> show g729
0/0 encoders/decoders of 4 licensed channels are currently in use
Asterisk1*CLI>


correcty me if i am wrong

right now i am on ztdummy
and i am able to make calls smoothly in maual mode

[asterisk_at_my_risk]

hello
i have taken a backup of my codec and as far as registration is consider
i fell that my codec are registered
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.12.1 currently running on Asterisk1 (pid = 2660)
Verbosity is at least 10
Asterisk1*CLI> show g729
0/0 encoders/decoders of 4 licensed channels are currently in use
Asterisk1*CLI>


correcty me if i am wrong

right now i am on ztdummy

i am able to make calls in manual mode smoothly (ie one to one calling)

[rudra_ach]

Hi,
Its not really true what you said the SIP provider at India. I have recently setup a vicidial at our India Office remotely from USA. Working with very charm with little memory release issue. Let me know who is your provider, I can share my configuration. I have tested for sify, premius, HCL, VSNL. All this SIP provider work great with licensed g729.

Coming to your free g729, it’s not working with recent version of Asterisk.

Here is my best suggestion. As you said, you have tried a lot to resolve it.

Have two server ready. One with Vicidial to generate call and other to receive the call. Then just play around to pin down, what exactly the cause.
From the same problem server try to send traffic with ulaw or alaw or GSM and see the exact issue.

Also one more thing, just follow the steps exactly mentioned, while installing and do a reinstall.

I was facing the same problem like Work in manual dial and not in auto dial. But lucky to pin down the problem in 3 hours, and found I missed the tT option while transferring the call.

Regards.
Rudra.

[asterisk_at_my_risk]

Hello all

i have been in this issue for a long time i was using g729 but finaly i got a vendor in india who was able to terminate my calls on g711 but to my surprize i saw same issue of warning in auto mode of calls in g711 also
i feel now its not a issue of g729 now please suggest something so that i can get out of this vicious cycle

my asterisk version is 1.2.12.1
my linux distro is FC4
my astguicleint version is 2.0.1

here is logs from the auto calls
***********************************************************
To: <sip:3003@192.168.6.15:6234>;tag=4c1f7766
From: "S0611220204378600051"<sip:asterisk@203.122.26.232>;tag=as0e1d50cd
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK2ffa54bd;rport=5060;received=203.122.26.232
Call-ID: 68e969987947ec8e5c8187a816dda1a3@203.122.26.232
CSeq: 102 INVITE
Contact: <sip:3003@192.168.6.15:6234>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: 232

v=0
o=- 35612219 35612510 IN IP4 192.168.6.15
s=eyeBeam
c=IN IP4 192.168.6.15
t=0 0
m=audio 6182 RTP/AVP 8 0 101
a=alt:1 1 : 2EC32F6F 000000A5 192.168.6.15 6182
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

--- (10 headers 10 lines)---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.6.15:6182
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:3003@192.168.6.15:6234>
set_destination: Parsing <sip:3003@192.168.6.15:6234> for address/port to send to
set_destination: set destination to 192.168.6.15, port 6234
Transmitting (NAT) to 203.122.26.230:6234:
ACK sip:3003@192.168.6.15:6234 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK66d5e04b;rport
From: "S0611220204378600051" <sip:asterisk@203.122.26.232>;tag=as0e1d50cd
To: <sip:3003@192.168.6.15:6234>;tag=4c1f7766
Contact: <sip:asterisk@203.122.26.232>
Call-ID: 68e969987947ec8e5c8187a816dda1a3@203.122.26.232
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
> Channel SIP/3003-08ce8a48 was answered.
-- Executing MeetMe("SIP/3003-08ce8a48", "8600051") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
Destroying call '57227ee03817194e4d6bacc241aeed83@127.0.0.1'
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/99912127773456@default-3e81,2", "call_log.agi|99912127773456") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial("Local/99912127773456@default-3e81,2", "sip/9912127773456@SIPTRUNK|55|tTo") in new stack
We're at 203.122.26.232 port 19568
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 209.189.127.36:5060:
INVITE sip:9912127773456@209.189.127.36 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK23e8f018;rport
From: "V1122020434000000001" <sip:2132689495@203.122.26.232>;tag=as23c81a71
To: <sip:9912127773456@209.189.127.36>
Contact: <sip:2132689495@203.122.26.232>
Call-ID: 5a2240dd70c19e41538724656730302c@203.122.26.232
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 21 Nov 2006 17:04:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 6750 6750 IN IP4 203.122.26.232
s=session
c=IN IP4 203.122.26.232
t=0 0
m=audio 19568 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 9912127773456@SIPTRUNK

<-- SIP read from 209.189.127.36:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK23e8f018;rport
From: "V1122020434000000001" <sip:2132689495@203.122.26.232>;tag=as23c81a71
To: <sip:9912127773456@209.189.127.36>
Call-ID: 5a2240dd70c19e41538724656730302c@203.122.26.232
CSeq: 102 INVITE
Content-Length: 0


--- (7 headers 0 lines)---

<-- SIP read from 209.189.127.36:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK23e8f018;rport
From: "V1122020434000000001" <sip:2132689495@203.122.26.232>;tag=as23c81a71
To: <sip:9912127773456@209.189.127.36>
Call-ID: 5a2240dd70c19e41538724656730302c@203.122.26.232
CSeq: 102 INVITE
Content-Length: 0


--- (7 headers 0 lines)---

<-- SIP read from 209.189.127.36:5060:
SIP/2.0 183 Session Progress
From: "V1122020434000000001" <sip:2132689495@203.122.26.232>;tag=as23c81a71
To: <sip:9912127773456@209.189.127.36>;tag=SD8g88699-0dd40708
Call-ID: 5a2240dd70c19e41538724656730302c@203.122.26.232
CSeq: 102 INVITE
Allow: OPTIONS,INVITE,CANCEL,ACK,BYE,PRACK,INFO
Content-Disposition: session;handling=required
Content-Type: application/sdp
Contact: <sip:12127773456@209.189.127.36:5060>
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK23e8f018;rport
Content-Length: 168

v=0
o=Sonus_UAC 9677 15903 IN IP4 216.112.169.44
s=SIP Media Capabilities
c=IN IP4 216.112.169.44
t=0 0
m=audio 25322 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=sendrecv


--- (11 headers 9 lines)---
Found RTP audio format 8
Peer audio RTP is at port 216.112.169.44:25322
Found description format PCMA
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
-- SIP/SIPTRUNK-08cf4378 is making progress passing it to Local/99912127773456@default-3e81,2

<-- SIP read from 209.189.127.36:5060:
SIP/2.0 200 OK
From: "V1122020434000000001" <sip:2132689495@203.122.26.232>;tag=as23c81a71
To: <sip:9912127773456@209.189.127.36>;tag=SD8g88699-0dd40708
Call-ID: 5a2240dd70c19e41538724656730302c@203.122.26.232
CSeq: 102 INVITE
Accept: multipart/mixed,application/sdp,application/isup,application/dtmf,application/dtmf-relay
Allow: OPTIONS,INVITE,CANCEL,ACK,BYE,PRACK,INFO
Content-Disposition: session;handling=required
Content-Type: application/sdp
Contact: <sip:12127773456@209.189.127.36:5060>
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK23e8f018;rport
Content-Length: 168

v=0
o=Sonus_UAC 9677 15903 IN IP4 216.112.169.44
s=SIP Media Capabilities
c=IN IP4 216.112.169.44
t=0 0
m=audio 25322 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=sendrecv


--- (12 headers 9 lines)---
Found RTP audio format 8
Peer audio RTP is at port 216.112.169.44:25322
Found description format PCMA
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
list_route: hop: <sip:12127773456@209.189.127.36:5060>
set_destination: Parsing <sip:12127773456@209.189.127.36:5060> for address/port to send to
set_destination: set destination to 209.189.127.36, port 5060
Transmitting (no NAT) to 209.189.127.36:5060:
ACK sip:12127773456@209.189.127.36:5060 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK6154008c;rport
From: "V1122020434000000001" <sip:2132689495@203.122.26.232>;tag=as23c81a71
To: <sip:9912127773456@209.189.127.36>;tag=SD8g88699-0dd40708
Contact: <sip:2132689495@203.122.26.232>
Call-ID: 5a2240dd70c19e41538724656730302c@203.122.26.232
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/SIPTRUNK-08cf4378 answered Local/99912127773456@default-3e81,2
> Channel Local/99912127773456@default-3e81,1 was answered.
-- Executing AGI("Local/99912127773456@default-3e81,1", "call_log.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
== Spawn extension (default, 99912127773456, 2) exited non-zero on 'Local/99912127773456@default-3e81,2'
-- Executing DeadAGI("Local/99912127773456@default-3e81,2", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing AGI("SIP/SIPTRUNK-08cf4378", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("Local/99912127773456@default-3e81,2", "VD_hangup.agi|PRI-----NODEBUG-----16-----ANSWER-----1-----0") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing MeetMe("SIP/SIPTRUNK-08cf4378", "8600051") in new stack
-- AGI Script VD_hangup.agi completed, returning 0
Nov 22 02:04:36 WARNING[7081]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/64)
Nov 22 02:04:36 WARNING[7081]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/64)
Nov 22 02:04:36 WARNING[7081]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/64)
Nov 22 02:04:36 WARNING[7081]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/64)
ov 22 02:04:50 WARNING[7081]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/64)
Nov 22 02:04:49 WARNING[7081]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/64)
Nov 22 02:04:50 WARNING[7081]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/64)
Nov 22 02:04:50 WARNING[7081]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/64)
Nov 22 02:04:50 WARNING[7081]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/64)
Nov 22 02:04:50 WARNING[7081]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/64)
Nov 22 02:04:50 WARNING[7081]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/64)
Nov 22 02:04:50 WARNING[7081]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/64)
Nov 22 02:04:50 WARNING[7081]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/64)
Nov 22 02:04:50 WARNING[7081]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/64)
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/SIPTRUNK-08cf4378'
-- Executing DeadAGI("SIP/SIPTRUNK-08cf4378", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("SIP/SIPTRUNK-08cf4378", "VD_hangup.agi|PRI-----NODEBUG-----0---------------") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
Scheduling destruction of call '5a2240dd70c19e41538724656730302c@203.122.26.232' in 32000 ms
set_destination: Parsing <sip:12127773456@209.189.127.36:5060> for address/port to send to
set_destination: set destination to 209.189.127.36, port 5060
Reliably Transmitting (no NAT) to 209.189.127.36:5060:
BYE sip:12127773456@209.189.127.36:5060 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK092bdc4c;rport
From: "V1122020434000000001" <sip:2132689495@203.122.26.232>;tag=as23c81a71
To: <sip:9912127773456@209.189.127.36>;tag=SD8g88699-0dd40708
Contact: <sip:2132689495@203.122.26.232>
Call-ID: 5a2240dd70c19e41538724656730302c@203.122.26.232
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
*************************************************************


give us some help or some clue

[mflorell]

What codec is the agent using?

Have you tried upgrading to 1.2.13 or downgrading to 1.2.10?

[asterisk_at_my_risk]

thanks matt for your prompt reply

i am using eyebeam paid version currently i have enabled g711alaw
and g711ulaw on my client

and i have not yet treid with and other version of asterisk if you recommed i can try on some other version also but will the zaptel and lipbri version along with the patch will remain same

[mflorell]

current zaptel and libpri should work for 1.2.10 and 1.2.13

[asterisk_at_my_risk]

sorry to bother you again but this one is very critical for us and i think that i am going wrong over on patches

i am going to work with asterisk 1.2.13 for testing
there are 3 patches as per scrach install

Patch 1(optional)
(1.2 tree) If you want to include Answering Machine Detection ability

Patch 2
(1.2 tree) apply the cli delimiter patch
cli_chan_concise_delimiter.patch

Patch 3
channel.c-42600.patch

do i have to apply al of these three patches in the case of 1.2.13 ?

and with all due respect sir actualy what is the reason for these type of error i realy want to know the root cause for this issue i feel now it is noe codec specific

thanks again fopr your kind help

[mflorell]

cli_chan_concise_delimiter.patch is the only required patch.

What kind of zaptel timer are you using?

[asterisk_at_my_risk]

hello matt

we are using ztdummy for the timer

[mflorell]

What is the ztdummy timing source?

What linux kernel version are you using?

[asterisk_at_my_risk]

our linux version

cat /proc/version
Linux version 2.6.11-1.1369_FC4 (bhcompile@decompose.build.redhat.com) (gcc version 4.0.0 20050525 (Red Hat 4.0.0-9)) #1 Thu Jun 2 22:55:56 EDT 2005

What is the ztdummy timing source?
well honestly i am not sure on this
what we have done is we have executed these two commands

modprobe zaptel
modprobe ztdummy

i read from voip-info that

" On kernel version 2.6 ztdummy uses internal high-resolution kernel timer and does not require any USB. Using the internal kernel timer is recommended."
when i run my lsmod command i get this
[root@phone ~]# lsmod
Module Size Used by
ztdummy 3924 0
zaptel 210820 5 ztdummy
crc_ccitt 2113 1 zaptel
md5 4033 1
ipv6 268097 18
parport_pc 28933 1
lp 13001 0
parport 40585 2 parport_pc,lp
autofs4 29253 2
rfcomm 42333 0
l2cap 30661 5 rfcomm
bluetooth 56133 4 rfcomm,l2cap
sunrpc 167813 1


i fell that we are wrong somewhere on zapata settings


we have done no configuration on zapata.conf its on default state

[mflorell]

I believe that the kernel timer frequency defaults to 100HZ which can cause problems.

From what I have read on several audio-related boards, you should change this to 1000HZ and rebuild your kernel. This is something that requires you to recompile the kernel, but it can have a large effect on audio delivery and quality.