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New Member

calling sip line from outside

dears,

any one can help, i have these config in my translation rules

incoming calls are working when i dial 2837555 from outside , but some are saying that when they call from certain numbers the number 2837555 it will be busy.

                 

!

voice translation-rule 1

rule 1 /2837599/ /599/

rule 2 /2837595/ /595/

rule 3 /2837597/ /597/

rule 4 /2837598/ /598/

rule 5 /2837400/ /100/

rule 6 /2837596/ /596/

rule 7 /.*2837555/ /123/

rule 8 /^1\(.......$\)/ /9\1/

rule 9 /\(^5........$\)/ /90\1/

rule 10 /^9\(.*\)/ /\1/

rule 11 /^.*\(123\)/ /\1/

!

voice translation-rule 191

rule 1 /^599/ /2837599/

rule 2 /^596/ /2837596/

rule 3 /^595/ /2837595/

rule 4 /^597/ /2837597/

rule 5 /^598/ /2837598/

rule 6 /^3....\*\(.\)/ /\1/

rule 7 /^9\(.*\)/ /\1/

rule 8 /.../ /2837555/

!

!

voice translation-profile SIP-IN

translate calling 1

translate called 1

!

voice translation-profile TP_OUT_SIP

translate calling 191

translate called 191

!

Everyone's tags (4)
24 REPLIES
VIP Super Bronze

calling sip line from outside

can you send a debug ccsip messages and debug voip ccapi inout

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New Member

Re: calling sip line from outside

hi,

tthanks for your support, please find logs as requested , called from 2835625 to 2837555 but call not going to AA

VIP Super Bronze

Re: calling sip line from outside

Hi, here is an analysis of your trace..

+++Your sip provider sends an invite without SDP+++So you are doing delay offer

Received:

INVITE sip:2837555@10.196.106.122:5060;

user=phone SIP/2.0 Via: SIP/2.0/UDP 10.200.7.157:5060;

branch=z9hG4bKj8nb8liui7vhlohlv7bnvkhkc

Call-ID: SBCk2aedpp2u2otuptbbfes2okad7aehaec@SoftX3000

From: <12835625>;tag=tpobhu4b-CC-36

To: <2837555>

+++After a couple of trying and invite to CUE, you send a 200 ok to provider with SDP++++.

Sent:
SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.7.157:5060;
branch=z9hG4bKj8nb8liui7vhlohlv7bnvkhkc
From: <12835625>;tag=tpobhu4b-CC-36
To: <2837555>;tag=2E4C4AD0-263D Date: Sat, 02 Jun 2012 08:49:51 GMT
Call-ID: SBCk2aedpp2u2otuptbbfes2okad7aehaec@SoftX3000
CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events:
telephone-event Remote-Party-ID: <123>;party=called;screen=no;privacy=off
Contact: <2837555> Supported: replaces Call-Info: <10.196.106.122:5060>;method="NOTIFY;Event=telephone-event;Duration=2000" Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x Supported: timer Content-Type: application/sdp
Content-Disposition: session;handling=required Content-Length: 251 
v=0
o=CiscoSystemsSIP-GW-UserAgent 336 981
IN IP4 10.196.106.122
s=SIP Call
c=IN IP4 10.196.106.122
t=0 0
m=audio 17614 RTP/AVP 0 101
c=IN IP4 10.196.106.122
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15 a=ptime:20

++++++++Now your provider sends an ACK without sdp+++++++++++ (for it to work they need to send an ACK with SDP)

Received:
ACK sip:2837555@10.196.106.122:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.7.157:5060;
branch=z9hG4bKeljk7bluenjvkjibkvniyujjx
Call-ID: SBCk2aedpp2u2otuptbbfes2okad7aehaec@SoftX3000
From: <12835625>;tag=tpobhu4b-CC-36
To: <2837555>;tag=2E4C4AD0-263D CSeq: 1
ACK Max-Forwards: 70 Content-Length: 0

+++Then your provider sends a bye+++++++++++

Received:
BYE sip:2837555@10.196.106.122:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKxnknekbionkvbvoxbnkliiexc
Call-ID: SBCk2aedpp2u2otuptbbfes2okad7aehaec@SoftX3000
From: <12835625>;tag=tpobhu4b-CC-36
To: <2837555>;tag=2E4C4AD0-263D
CSeq: 2 BYE Max-Forwards: 70
Reason: Q.850;cause=41;text="temporary failure"
Content-Length: 0 

+++The ccapi cause code helps us more++++

001281: Jun  2 08:49:51.877: //127822/C058B05B8462/CCAPI/cc_api_call_disconnected:
   Cause Value=96, Interface=0x8738CB14, Call Id=127822

Cause 96 means that a mandatory information element is missing..Now the question is what is the mandatory information your provider is expecting in the 200 answer that you didnt provide..and I think its this sip session attribute

a=sendrecv or a=sendonly

Now according to RFC 3264  An Offer/Answer Model Session Description Protocol , it stipulates that

If the offerer wishes to both send and   receive media with its peer, it MAY include an "a=sendrecv"
   attribute, or it MAY omit it, since sendrecv is the default

However I think Huwaei your sip provider still want to see a session attribute in your offer. Hence the reason for disconnecting the call.

My suggestion is to use early offer from the provider. That way they can send you their attributes and ccme can just ack that.

It is my experience with this provider that they always send a session atrribute even if its  (a=sendrecv) which is the default.

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New Member

Re: calling sip line from outside

hi,

thanks for your support

i am not sur that i get you

can you specify what to do to solve this issue.

New Member

Re: calling sip line from outside

hi

you mean i have to configure early-offer as below

Configuring Delayed-Offer to Early-Offer for SIP Audio Calls at the Global Level

To configure Delayed-Offer to Early-Offer for SIP Audio Calls at the global level, perform the steps in this section.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice service voip

4. allow-connections sip

5. early-offer forced

6. exit

VIP Super Bronze

calling sip line from outside

Yes try that and test again...send the debugs again after you have configured this.

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New Member

Re: calling sip line from outside

hi, i tried to add the commands but it doesnt support

Re: calling sip line from outside

Try to create voice class sip-profile. Add the missing header in your 200 OK response and test.

voice class sip-profile 1

Response 200 sdp-header attribute add "sendrecv"

Then assign this profile to your matched dialpeer.

Baqari

Sent from Cisco Technical Support iPhone App

New Member

Re: calling sip line from outside

Hi Baqari,

i tried adding the profile, but the call is not going to autoattend, but it keeps on ringing, if i remove the command from the dial peer, the specific number is not working but other nos are going to the auto attend.

Re: calling sip line from outside

Please share the debig after adding the command

New Member

Re: calling sip line from outside

                 hi

debug as requested

VIP Super Bronze

Re: calling sip line from outside

Can you please add this..

voice class sip-profile 1

Response 200 sdp-header attribute add "a=sendrecv"

NB: That the previuos config didnt have a=sendrecv, so you provider was ignoring your 200 ok.

Ensure to apply the profile to the dial-peer going to your sip provider as you did before. Please test again and send debug ccsip messages

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Re: calling sip line from outside

Hi Edwin,

In fact its typo mistake I forgot to put "a=" while posting the SIP-Profile. Apologize for this. I can see the SDP header is added as "sendrecv" in your debugs without "a=".

I am pasting the new SIP-Profile with the correction.

voice class sip-profile 1

response 200 sdp-header attribute add "a=sendrecv".

Try now and paste the result.

Thanks to aokanlawon for picking it up

New Member

Re: calling sip line from outside

  hi,

still the same, it keeps on ringing without going to the AA        

VIP Super Bronze

Re: calling sip line from outside

Hi,

Did you remove the previous wrong sip profile config because I can still see the wrong attribute sent to your provider..The one underlined is still there and this format is wrong.

s=SIP Call

c=IN IP4 10.196.106.122

t=0 0

sendrecv

a=sendrecv

Please ensure you do not have:

voice class sip-profile 1

response 200 sdp-header attribute add "sendrecv"

response 200 sdp-header attribute add "a=sendrecv"

That the only one you have is :

voice class sip-profile 1

response 200 sdp-header attribute add "a=sendrecv"

Please rate useful posts

"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: calling sip line from outside

                   hi,

thanks for the support sorry to say it didnt work, all other calls goes to aa, but the making call as early specified no. keeps on ringing

VIP Super Bronze

calling sip line from outside

Edwin,

Can you confirm that this number is your AA number 2837555

Also remove the voice profile configuration because your provider doesnt like it and is not responding to it.

Can you also confirm you have other AA numbers that are working..Is this just the only number that is not workinmg?

If you have other AA number that is working can you send  a trace for a working one?

Please rate useful posts

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New Member

calling sip line from outside

hi ,

2837555 is the only number

aa extn is 123

all other nos are configured for directline

VIP Super Bronze

calling sip line from outside

Edwin,

I have seen a similar problem with this particular sip provider for calls to AA on unity express. To confirm if you are having the same problem can you tell me what codec you use for direct calls? is it G711alaw.. You can send a trace for calls to direct line here so I can quickly have a look.

Can you also configure a test call such that your direct line uses G711ulaw to the provider.  Let me knoe if you need help with the config.

The problem is similar to what a user on this forum has with this same provider for calls going to CUE using g711ulaw.

So if you can do this tests and confirm then we can determine if your issue is similar

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Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: calling sip line from outside

hi,

this is provided by the SIP line provider

         

Customer IP address= 10.196.106.122/30

STC IP Address =   10.196.106.121/30

Protocol= SIP

SIP Port = 5060

Transport Protocol=UDP

Voice Codec= G711 A-Law

DTMF = IN-Band DTMF with RFC2833

SIP Server IP address = 10.200.7.157

VIP Super Bronze

calling sip line from outside

So there is your problem..Your provider cant do G711ulaw. This is the only codec supported by Unity express. Have you confirmed that your phones are only doing G711alaw. This is important.

If this is the case you either speak to your provider or find another provider because calls to voicemail will not work and AA will not work.

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New Member

Re: calling sip line from outside

hi,

but i dont have problem with any other nos calling me.

i am not getting the calls from the number  to 2835625

when i dial from 2835625 to my number 2837555, it gives a busy tone.

VIP Super Bronze

calling sip line from outside

Edwin,

Can you please aswer my questions. What codecs do your direct calls use? G711alaw or G711ulaw? Please answer.

I dont know what numbers you are talking about, bear in mind I dont know what your system config is like.  You said 2837555 is AA (Auot attendant) How is it your number again...I am confused..

Please rate useful posts

"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: calling sip line from outside

hi,

can you please provide the config for test call such that your direct line uses G711ulaw to the provider.

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