Big problems during transfer

New install of the Distro and setup from scratch.
Using Digium Te122 to a PRI
Asterisk version 1.8.17.0
FreePBX version 1.817.210.58-1
I have FOP2 installed
Things seem to be going smoothly except for call transfers. And only sometimes. I couple of different things will happen when the transfer doesn’t go as it should. Sometimes the girl will hit transfer and will say transfer failed on her phone (cisco 7940G) and she will have to resume the call and do it again once or twice and it will finally go and other times she will transfer and it will fail and show blinking like they are still in limbo (hold) but she can hear the person in their office pick up and take the call even though she apparently still has them on hold at her phone. I was able to catch a failure and pasted it below. The line with the notice that says “Unable to create/find SIP channel for this INVITE” is where She says the transfer is failing. It seems to happen on regular and blind transfers as well. Any help would be appreciated and let me know anything you need to see and I can post it on here quickly. Thanks

– Executing [[email protected]:10] Set(“DAHDI/i1/2708328121-795”, “ITER=2”) in new stack
– Executing [[email protected]:11] GotoIf(“DAHDI/i1/2708328121-795”, “0?begin”) in new stack
– Executing [[email protected]:12] Set(“DAHDI/i1/2708328121-795”, “DSTRING=SIP/3503”) in new stack
– Executing [[email protected]:13] Return(“DAHDI/i1/2708328121-795”, “”) in new stack
– Executing [[email protected]:27] GotoIf(“DAHDI/i1/2708328121-795”, “0?nodial”) in new stack
– Executing [[email protected]:28] GotoIf(“DAHDI/i1/2708328121-795”, “0?skiptrace”) in new stack
– Executing [[email protected]:29] GosubIf(“DAHDI/i1/2708328121-795”, “1?ctset,1():ctclear,1()”) in new stack
– Executing [[email protected]:1] Set(“DAHDI/i1/2708328121-795”, “DB(CALLTRACE/3503)=2708328121”) in new stack
– Executing [[email protected]:2] Return(“DAHDI/i1/2708328121-795”, “”) in new stack
– Executing [[email protected]:30] Set(“DAHDI/i1/2708328121-795”, “D_OPTIONS=trM(auto-blkvm)”) in new stack
– Executing [[email protected]:31] ExecIf(“DAHDI/i1/2708328121-795”, “0?SIPAddHeader(Alert-Info: )”) in new stack
– Executing [[email protected]:32] ExecIf(“DAHDI/i1/2708328121-795”, “0?SIPAddHeader()”) in new stack
– Executing [[email protected]:33] ExecIf(“DAHDI/i1/2708328121-795”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [[email protected]:34] GosubIf(“DAHDI/i1/2708328121-795”, “0?qwait,1()”) in new stack
– Executing [[email protected]:35] Set(“DAHDI/i1/2708328121-795”, “__CWIGNORE=”) in new stack
– Executing [[email protected]:36] Set(“DAHDI/i1/2708328121-795”, “__KEEPCID=TRUE”) in new stack
– Executing [[email protected]:37] GotoIf(“DAHDI/i1/2708328121-795”, “0?usegoto,1”) in new stack
– Executing [[email protected]:38] GotoIf(“DAHDI/i1/2708328121-795”, “1?godial”) in new stack
– Goto (macro-dial-one,s,42)
– Executing [[email protected]:42] Dial(“DAHDI/i1/2708328121-795”, “SIP/3503,15,trM(auto-blkvm)”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/3503
– Connected line update to SIP/3569-00001862 prevented.
– SIP/3519-00001863 is ringing
– SIP/3503-00001864 is ringing
– Connected line update to SIP/3569-00001862 prevented.
– SIP/3519-00001863 answered SIP/3569-00001862
[2012-10-31 10:00:52] NOTICE[3281]: chan_sip.c:23253 handle_request_invite: Unable to create/find SIP channel for this INVITE
– SIP/3503-00001864 answered DAHDI/i1/2708328121-795
– Executing [[email protected]:1] Set(“SIP/3503-00001864”, “__MACRO_RESULT=”) in new stack
– Executing [[email protected]:2] Macro(“SIP/3503-00001864”, “blkvm-clr,”) in new stack
– Executing [[email protected]:1] Set(“SIP/3503-00001864”, “SHARED(BLKVM,DAHDI/i1/2708328121-795)=”) in new stack
– Executing [[email protected]:2] Set(“SIP/3503-00001864”, “GOSUB_RETVAL=”) in new stack
– Executing [[email protected]:3] MacroExit(“SIP/3503-00001864”, “”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/3503-00001864”, “0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=3503)”) in new stack
– Executing [[email protected]:4] ExecIf(“SIP/3503-00001864”, “0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Ollie Calhoun)”) in new stack

Any help?

What kind of router are you using? I get transfer issues with some routers. Some units whack the SIP headers.

Thank you for the response. Seems a bit tough getting good help. I have two 48 port Cisoc SF200 switches and a Cisco ASA5505 Firewall. Does that help you out any?

“Seems a bit tough getting good help.”

Feel like you’re drowning? Start breathing again and leave the coding to us… we’re the best when it comes to FreePBX and Asterisk.

follow the link and drop $129 for an hour. How much are you losing being unproductive? The ASA could cause problems. Cisco is very Asterisk friendly, but you have to make sure all the ports are opened up – 5060, 10000:20000 UDP. Forwarding all traffic in those UDP slots to your private LAN address of your server running FreePBX. Good luck. www.corysanders.com

The only reason I haven’t called in is because it seems like something so simple I just can’t put my finger on. The thing is, it worked in the beginning and started doing this about 10 days after a fresh install. Also, if it was a port that wasn’t opened then it would be consistent. If it doesn’t let the call pass through, I can pick it up another time or two and try it and it will eventually go through. I wouldn’t think that would be a port issue would you?

Anyone else have any ideas that could be helpful?

wrong with your router, I think. What kind of router is it? Make and model and how old?

I don’t see how it could be the router. I gave the info about it to ya above. Things were working fine through it before and there hasn’t been a single change to it in a year. It just doesn’t make since to me. Any other ideas? Thanks

Nobody in here here has any other ideas at all???

and drop $129 on an hour of service from the FreePBX team

How can anyone have any “other ideas”, your post finished with

  • Executing [[email protected]:3] ExecIf(“SIP/3503-00001864”, “0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=3503)”) in new stack
    – Executing [[email protected]:4] ExecIf(“SIP/3503-00001864”, “0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Ollie Calhoun)”) in new stack

That is normal and indicates an ongoing call, you need to actually post the failing part of your call.

nolos, If I were you, I would do everything dicko says. Everything. Listen and follow directions closely. He is and has offered to help you. He can. Go through this post and format a reply that is going to make him smile and he will continue on with you. Learned everything I know from the guy.

Exception 0045H - possible sarcasm detected in post

Alert created by Drupal pre-alpha neural management

When we transfer, sometimes we get a problem line in the asterisk CLI that is just above the bottom line of code that I posted. It is as follows:
[2012-10-31 10:00:52] NOTICE[3281]: chan_sip.c:23253 handle_request_invite: Unable to create/find SIP channel for this INVITE
When we get that line of code, the call fails to transfer. One of two things happen when we see that code. Either the phone (ciso 7940) physically displays Transfer Failed and we have to try it a couple more times and then it works. OR the line blinks as we are attempting to call the destination extension and once we hot transfer again to complete the call transfer, the source call just stays blinking like we never tried to transfer it. If the latter is the case then we have to Resume the call and try to transfer it again. If you need any other files then just let me know.

Is this problem replicated when you transfer using the feature code instead of the phone buttons or soft keys?

That’s a good question. I’m having Andrea transfer using ##extension all day today and she will let me know if the transfer fails. Thanks for the thoughts and I will let you know.

Seems like we are having some problems using the feature codes to transfer as well. I have noticed that when the ## (blind transfer) works there is still a long pause on the phone before you see the call go away. I have been keeping an eye on the code and this is the first error I saw pop up. Is it related?

[2012-11-20 11:51:48] NOTICE[10475]: features.c:3623 feature_request_and_dial: We exceeded our AT-timeout for Local/[email protected];1
== Spawn extension (macro-dial-one, s, 42) exited non-zero on ‘Local/[email protected];2’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 14) exited non-zero on ‘Local/[email protected];2’ in macro ‘exten-vm’
== Spawn extension (from-internal-xfer, 3533, 2) exited non-zero on ‘Local/[email protected];2’
– <SIP/3500-00006846> Playing ‘beep.ulaw’ (language ‘en’)
– Executing [[email protected]:1] Macro(“Local/[email protected];2”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“Local/[email protected];2”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [[email protected]:3] ExecIf(“Local/[email protected];2”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [[email protected]:4] Hangup(“Local/[email protected];2”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘Local/[email protected];2’ in macro ‘hangupcall’
== Spawn extension (from-internal-xfer, h, 1) exited non-zero on ‘Local/[email protected];2’
[2012-11-20 11:51:48] WARNING[10475]: features.c:2634 builtin_atxfer: Failed to play transfer sound!
– Stopped music on hold on DAHDI/i1/6189224169-212c
– Executing [[email protected]:1] Macro(“DAHDI/i1/6189224169-212c”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“DAHDI/i1/6189224169-212c”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [[email protected]:3] ExecIf(“DAHDI/i1/6189224169-212c”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [[email protected]:4] Hangup(“DAHDI/i1/6189224169-212c”, “”) in new stack

hey, reconwireless, does that help any?