calls not coming on softphone xlite

[iTGsoft]

Hi,

I am able to dial manual calls and can also do automatic dialing but the calls are not coming on softphone xlite.

can anybody help on this please?

[astgeeks solutions]

HI,

I guess u have an issue with audio codec.Pls check in ur sip configuration what codec u r using.if u r using g729 in ur sip config and try to take predictive calls or make manul calls from xlite.it will be not possible then .

[iTGsoft]

Thanks for the reply!

I am able to make manual calls as I said and even the live calls are also coming but that call is not coming on my soft phone xlite , is it dial plan related configuration issue?

when this situation occurs is it totally a voip min. provider issue because of which call is not throwing on softphone or something related to vicidial configuration

pls suggest.

[astgeeks solutions]

i don't understand one thing that u r able to make manual and predictive calls from where??is it from using audiocodec voip gateway??and when u r trying the same through xlite softphone it is not working for neither manual calls and not for predicitive also???

[iTGsoft]

I think I am not able to make it clear...well ..I am able to solve this step but I am stucked further. I am dialing from my softphone through asterisk using a vip account, call rings and softphone shows call established but no voice comes in headset and asterisk promt shows the error "Unable to find a codec translation path from g729 to ulaw".
and on softphone the codec g711u

pls help me on this..
Thanks

[gardo]

It would really help us a lot to know more about your system.

Asterisk version?
ViciDial version
Linux distro?

Post the output of the Asterisk CLI when dialing and that will help us to identify what's happening to your system.

[iTGsoft]

Asterisk is 1.2.30.4
we are using vicibox server
asterisk CLI says:----

-- Registered SIP 'cc100' at 192.168.10.16 port 8448 expires 3600
-- Saved useragent "CounterPath eyeBeam release 3014w stamp 25980" for peer cc100
-- Executing AGI("SIP/cc100-081cf2c0", "agi://127.0.0.1:4577/call_log") in new stack
Sep 2 09:53:21 WARNING[4809]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x817a728', 9 retries!
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc100-081cf2c0", "SIP/12127773020/12127773456||tTor") in new stack
-- [size=18]Called 12127773020/12127773456
Sep 2 09:53:23 WARNING[4820]: channel.c:2403 [b]set_format: Unable to find a codec translation path from g729 to ulaw
Sep 2 09:53:23 WARNING[4820]: channel.c:2403 set_format: Unable to find a codec translation path from g729 to ulawSIP/121277730200820bf18 is making progress passing it to SIP/cc100-081cf2c0
-- SIP/12127773020-0820bf18 answered SIP/cc100-081cf2c0
Sep 2 09:53:23 NOTICE[5569]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.10.16
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:23 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4)
Sep 2 09:53:24 WARNING[5569]: chan_sip.c:2608 sip_write: Ask


Thanks

[astgeeks solutions]

i have told u before that the problem is due to ur codec mismatch between softphone and ur asterisk sip configuration.

do one thing change ur sip config of cc100 to

disallow=all
allow=ulaw
allow=alaw

and change your voip minute account with the same.

disallow=all
allow=ulaw
allow=alaw

and u r done.

Now u r able to do calls manually and predictively from xlite softphone.
and dont change any codec settings into xlite.

[gardo]

Yep. Astgeeks solutions is right. Based on your Asterisk CLI, you have a codec issue. You're provider is using g729 codec. You need to have that codec installed on your Asterisk server. If not, just use ulaw or g711.

[iTGsoft]

Thanks
I am able to call from soft phone manually but when I try with vicidial using Agent GUI by pressing "dial next Number" , it picks up the number and asterisk CLI shows:-


== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/8600051@default-be13,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-be13,2", "SIP/12127773020/8600051||tTor") in new stack
-- Called 12127773020/8600051
Sep 2 17:28:53 WARNING[13714]: chan_sip.c:9890 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"M0902172833000061108" <sip:12127773020@64.56.64.110>;tag=as3447cc68'
-- SIP/12127773020-08202df0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("Local/8600051@default-be13,2", "") in new stack
== Spawn extension (default, 8600051, 3) exited non-zero on 'Local/8600051@default-be13,2'
-- Executing DeadAGI("Local/8600051@default-be13,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing Dial("Local/8600051@default-be13,2", "SIP/12127773020/h||tTor") in new stack
-- Called 12127773020/h
-- Got SIP response 604 "Does Not Exist Anywhere" back from 64.56.64.110
== No one is available to answer at this time (1:0/0/0)
-- Executing Hangup("Local/8600051@default-be13,2", "") in new stack
== Spawn extension (default, h, 3) exited non-zero on 'Local/8600051@default-be13,2'
-- Executing AGI("OutgoingSpoolFailed", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("OutgoingSpoolFailed", "SIP/12127773020/failed||tTor") in new stack
Sep 2 17:28:54 WARNING[15599]: channel.c:2594 ast_request: No translator path exists for channel type SIP (native 65535) to 0
Sep 2 17:28:54 NOTICE[15599]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'SIP' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("OutgoingSpoolFailed", "") in new stack
== Spawn extension (default, failed, 3) exited non-zero on 'OutgoingSpoolFailed'
-- Executing DeadAGI("OutgoingSpoolFailed", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing Dial("OutgoingSpoolFailed", "SIP/12127773020/h||tTor") in new stack
Sep 2 17:28:54 WARNING[15599]: channel.c:2594 ast_request: No translator path exists for channel type SIP (native 65535) to 0
Sep 2 17:28:54 NOTICE[15599]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'SIP' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("OutgoingSpoolFailed", "") in new stack
== Spawn extension (default, h, 3) exited non-zero on 'OutgoingSpoolFailed'



and nothing comes on softphone.
Can you pls guide me further?

[astgeeks solutions]

It looks something authentication problem with your voip account.i suggest to use a userbased voip account.that will solve ur problem.and pls send me a screen shot of asterisk cli using command 'sip debug'.

[iTGsoft]

Below is what my CLI says when using sip debug


12 headers, 0 lines
Reliably Transmitting (NAT) to 64.56.64.110:5060:
OPTIONS sip:64.56.64.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK2a0d466d;rport
From: "asterisk" <sip:asterisk@192.168.10.200>;tag=as574a8938
To: <sip:64.56.64.110>
Contact: <sip:asterisk@192.168.10.200>
Call-ID: 49084fd22852ad844f2ac2f365f65a30@192.168.10.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 02 Sep 2009 12:29:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---

<-- SIP read from 64.56.64.110:5060:
SIP/2.0 200 OK
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK2a0d466d
From: "asterisk" <sip:asterisk@192.168.10.200>;tag=as574a8938
Call-ID: 49084fd22852ad844f2ac2f365f65a30@192.168.10.200
To: <sip:64.56.64.110>;tag=020929090556
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Length: 0


--- (8 headers 0 lines) ---
Destroying call '49084fd22852ad844f2ac2f365f65a30@192.168.10.200'

<-- SIP read from 192.168.10.16:8448:

what is ---Spawn extension (default, h, 3) exited non-zero on 'OutgoingSpoolFailed'
I find this 'h' coming many times in previous CLI but when I try manually with softphone I find number instead of h

and what do u mean by user based voip account.

Thanks

[astgeeks solutions]

userbased account means there should be a username and password based account instead of registerd ip account.

[iTGsoft]

Thats what I understand from it thats the account I am using, still it gives the authentication issue and it comes only when dialing from vicidial, not at the time when I manually dial from soft phone using same account

any idea of this 'h' coming in logs
Any other clu pls?

[astgeeks solutions]

for your h configuration please follow vicidial managers manual..